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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-04 17:50:05 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-04 17:50:05 +0000
commitcc873a6344a045d2a54ac7b47f077134d80b356c (patch)
tree5bb19f215244c4cca37f42ec450398e21c6c4365
parent45b4d7c2b9ba25db60c9820e3483832cf42e2dca (diff)
Importing files for 1.6.0-beta5 release
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0-beta5@105665 f38db490-d61c-443f-a65b-d21fe96a405b
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diff --git a/.version b/.version
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+1.6.0-beta5
diff --git a/ChangeLog b/ChangeLog
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+2008-03-04 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-beta5 released.
+
+2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Update CHANGES heading
+
+ * funcs/func_version.c: Simplify a trivial snprintf() with
+ ast_copy_string()
+
+ * main/hashtab.c: Make it so you don't have to cast away const in a
+ couple places
+
+ * main/hashtab.c: remove unnecessary casts
+
+ * /: Blocked revisions 105591 via svnmerge ........ r105591 |
+ russell | 2008-03-03 22:31:29 -0600 (Mon, 03 Mar 2008) | 4 lines
+ Backport a minor bug fix from trunk that I found while doing
+ random code cleanup. Properly break out of the loop when a
+ context isn't found when verify that includes are valid. ........
+
+ * main/pbx.c: - Add curly braces around the while loop - Properly
+ break out of the loop on error when an included context is not
+ found
+
+ * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use
+ sizeof() instead of a magic number
+
+ * channels/chan_zap.c: Fix some code that was improperly changed in
+ revision 104866 from issue #12079. (closes issue #12129, reported
+ by elguero, patched by me)
+
+2008-03-03 18:08 +0000 [r105573] Jason Parker <jparker@digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) |
+ 7 lines Fix types for astNumChannels and astConfigCallsProcessed.
+ (closes issue #12114) Reported by: jeffg Patches: 12114.patch
+ uploaded by jeffg (license 192) ........
+
+2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 105570 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03
+ Mar 2008) | 3 lines In the case of an ast_channel allocation
+ failure, take the local_pvt out of the pvt list before destroying
+ it. ........
+
+ * channels/chan_local.c, /: Merged revisions 105568 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03
+ Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt
+ struct when ast_channel allocation fails. Also, in passing,
+ centralize the code necessary to destroy a local_pvt. ........
+
+ * main/autoservice.c, /: Merged revisions 105565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008)
+ | 3 lines Update the copyright information for autoservice. Most
+ of the code in this file now is stuff that I have written
+ recently ... ........
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/_private.h, main/asterisk.c: 3) In addition to
+ merging the changes below, change trunk back to a regular LIST
+ instead of an RWLIST. The way this list works makes it such that
+ a RWLIST provides no additional benefit. Also, a mutex is needed
+ for use with the thread condition. Merged revisions 105563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008)
+ | 24 lines Merge in some changes from
+ team/russell/autoservice-nochans-1.4 These changes fix up some
+ dubious code that I came across while auditing what happens in
+ the autoservice thread when there are no channels currently in
+ autoservice. 1) Change it so that autoservice thread doesn't keep
+ looping around calling ast_waitfor_n() on 0 channels twice a
+ second. Instead, use a thread condition so that the thread
+ properly goes to sleep and does not wake up until a channel is
+ put into autoservice. This actually fixes an interesting bug, as
+ well. If the autoservice thread is already running (almost always
+ is the case), then when the thread goes from having 0 channels to
+ have 1 channel to autoservice, that channel would have to wait
+ for up to 1/2 of a second to have the first frame read from it.
+ 2) Fix up the code in ast_waitfor_nandfds() for when it gets
+ called with no channels and no fds to poll() on, such as was the
+ case with the previous code for the autoservice thread. In this
+ case, the code would call alloca(0), and pass the result as the
+ first argument to poll(). In this case, the 2nd argument to
+ poll() specified that there were no fds, so this invalid pointer
+ shouldn't actually get dereferenced, but, this code makes it
+ explicit and ensures the pointers are NULL unless we have valid
+ data to put there. (related to issue #12116) ........
+
+2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 105560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7
+ lines It is possible for no audio to pass between the current
+ digit and next digit so expand logic that clears emulation to
+ AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg
+ Patches: v1-11911.patch uploaded by dimas (license 88) Tested by:
+ tbsky ........
+
+ * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6
+ lines Add a comment to describe some logic. (closes issue #12120)
+ Reported by: flefoll Patches:
+ chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
+ 244) ........
+
+2008-03-01 03:59 +0000 [r105509] Joshua Colp <jcolp@digium.com>
+
+ * main/slinfactory.c: Add support for 16KHz signed linear.
+
+2008-03-01 02:03 +0000 [r105479] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Drop bad property
+
+2008-03-01 01:30 +0000 [r105477] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, include/asterisk/app.h,
+ main/global_datastores.c, /, main/features.c, main/app.c,
+ include/asterisk/global_datastores.h: Asterisk, when parking can
+ drop rights a caller when a parking timeout occurs. Also, when
+ doing built-in attended transfers, sometimes incorrectly passes
+ rights from the transferrer to the transferee. This patch tries
+ to fixes the parking issue and lays some groundwork for later
+ fixing the transfer issue. (closes issue #11520) Reported by:
+ pliew Tested by: otherwiseguy
+
+2008-03-01 00:53 +0000 [r105461] Russell Bryant <russell@digium.com>
+
+ * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI
+ command to control custom device states. Also, do some additional
+ code cleanup and improvement in passing. (closes issue #12106)
+ Reported by: nizon Patches: devstate-patch.txt uploaded by nizon
+ (license 415) -- Updated to trunk, and tab completion added by me
+
+2008-02-29 23:53 +0000 [r105411] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Convert to use ast_str
+
+2008-02-29 23:36 +0000 [r105410] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 105409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008)
+ | 23 lines Fix a major bug in autoservice. There was a race
+ condition in the handling of the list of channels in autoservice.
+ The problem was that it was possible for a channel to get removed
+ from autoservice and destroyed, while the autoservice thread was
+ still messing with the channel. This led to memory corruption,
+ and caused crashes. This explains multiple backtraces I have seen
+ that have references to autoservice, but do to the nature of the
+ issue (memory corruption), could cause crashes in a number of
+ areas. (fixes the crash in BE-386) (closes issue #11694) (closes
+ issue #11940) The following issues could be related. If you are
+ the reporter of one of these, please update to include this fix
+ and try again. (potentially fixes issue #11189) (potentially
+ fixes issue #12107) (potentially fixes issue #11573) (potentially
+ fixes issue #12008) (potentially fixes issue #11189) (potentially
+ fixes issue #11993) (potentially fixes issue #11791) ........
+
+2008-02-29 18:34 +0000 [r105378] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample: Add documentation for setting
+ username/password in SIP dial string. (closes issue #11587)
+ Reported by: sobomax Patches: dialstring_doc.diff uploaded by
+ sobomax (license 359)
+
+2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008)
+ | 1 line Fix a potential memory leak ........
+
+ * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c:
+ Remove unnecessary if statements before calling iks_delete
+ (redundant check is done inside iks_delete), thus making the code
+ conform with coding guidelines.
+
+2008-02-29 13:55 +0000 [r105262] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb
+ 2008) | 4 lines Bump up the size of the uniqueid variable.
+ (closes issue #12107) Reported by: asgaroth ........
+
+2008-02-29 13:12 +0000 [r105210] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Automatically create new buddy upon reception
+ of a presence stanza of type subscribed. (closes issue #12066)
+ Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
+ phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
+ (license 73) Tested by: ffadaie, phsultan
+
+2008-02-29 01:15 +0000 [r105176] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008)
+ | 7 lines Update init script for LSB compat (closes issue #9843)
+ Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by
+ ibc (license 211) Tested by: paravoid ........
+
+2008-02-28 22:39 +0000 [r105144] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c:
+ Merged revisions 105116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008)
+ | 8 lines Fix a bug in the lock tracking code that was discovered
+ by mmichelson. The issue is that if the lock history array was
+ full, then the functions to mark a lock as acquired or not would
+ adjust the stats for whatever lock is at the end of the array,
+ which may not be itself. So, do a sanity check to make sure that
+ we're updating lock info for the proper lock. (This explains the
+ bizarre stats on lock #63 in BE-396, thanks Mark!) ........
+
+2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb
+ 2008) | 6 lines When using autofill, members who are in use
+ should be counted towards the number of available members to call
+ if ringinuse is set to yes. Thanks to jmls who brought this issue
+ up on IRC ........
+
+ * main/dial.c, /: Merged revisions 104841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb
+ 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels
+ a lockable list. This is because in some cases, the ast_dial may
+ exist in multiple threads due to asynchronous execution of its
+ application, and I found some cases where race conditions could
+ exist. 2. Check in ast_dial_join to be sure that the channel
+ still exists before attempting to lock it, since it could have
+ gotten hung up but the is_running_app flag on the
+ ast_dial_channel may not have been cleared yet. (closes issue
+ #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
+ putnopvut (license 60) Tested by: jvandal ........
+
+2008-02-28 19:21 +0000 [r105006] Jason Parker <jparker@digium.com>
+
+ * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb
+ 2008) | 9 lines Make pbx_exec pass an empty string into
+ applications, if we get NULL. This protects against possible
+ segfaults in applications that may try to use data before
+ checking length (ast_strdupa'ing it, for example) (closes issue
+ #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
+ uploaded by qwell (license 4) ........
+
+2008-02-28 14:42 +0000 [r104974] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc: Fix crash when configuration does not match
+ hardware detection. (closes issue #12096) Reported by: mmickan
+ Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
+
+2008-02-28 04:37 +0000 [r104921] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb
+ 2008) | 2 lines According to a video at www.cisco.com, the 7921G
+ supports 6 line appearances. ........
+
+2008-02-28 00:11 +0000 [r104869] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, build_tools/strip_nonapi: Merged revisions
+ 104868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008)
+ | 7 lines Compatibility fix for PPC64 (closes issue #12081)
+ Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch
+ uploaded by jcollie (license 412) Tested by: jcollie, Corydon76
+ ........
+
+2008-02-27 23:58 +0000 [r104866] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: reduce indentation in alloc_sub (issue
+ #12079) Reported by: tzafrir Patches: alloc_sub uploaded by
+ tzafrir (license 46)
+
+2008-02-27 21:02 +0000 [r104788] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2
+ lines Don't loop around infinitely trying to spy on our own
+ channel, and don't forget to free/detach the datastore upon
+ hangup of the spy. ........
+
+2008-02-27 20:37 +0000 [r104784] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/file.c: Merged revisions 104783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb
+ 2008) | 4 lines Bump a couple of more buffers up by 2 so that
+ annoying warnings aren't generated like crazy on every
+ fileexists_core call. ........
+
+2008-02-27 19:36 +0000 [r104756] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in
+ favor of 'val', which serves the exact same purpose.
+
+2008-02-27 18:20 +0000 [r104705] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 104704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008)
+ | 2 lines Ensure the session ID can't be 0. ........
+
+2008-02-27 17:45 +0000 [r104687] Joshua Colp <jcolp@digium.com>
+
+ * /, main/file.c: Merged revisions 104665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2
+ lines Bump up the buffer by 2. ........
+
+2008-02-27 17:36 +0000 [r104643] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008)
+ | 4 lines Fix a problem in ChanSpy where it could get stuck in an
+ infinite loop without being able to detect that the calling
+ channel hung up. (closes issue #12076, reported by junky, patched
+ by me) ........
+
+2008-02-27 17:31 +0000 [r104617] Jason Parker <jparker@digium.com>
+
+ * /, main/features.c: Merged revisions 104598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) |
+ 8 lines Inherit language from the transfering channel on a blind
+ transfer. (closes issue #11682) Reported by: caio1982 Patches:
+ local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22)
+ Tested by: caio1982, victoryure ........
+
+2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp <jcolp@digium.com>
+
+ * /, main/loader.c: Merged revisions 104596 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4
+ lines Use the lock (which already existed, it just wasn't used)
+ on the updaters list to protect the contents instead of the
+ overall module list lock. (closes issue #12080) Reported by:
+ ChaseVenters ........
+
+ * channels/chan_sip.c: After further discussion revert my previous
+ commit for this. Currently in order to ensure devicestate is the
+ expected value in another module (such as app_queue) then
+ chan_sip must be loaded before hand.
+
+2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/file.c: Merged revisions 104593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb
+ 2008) | 8 lines fallback to standard English prompts properly
+ when using new prompt directory layout (closes issue #11831)
+ Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG
+ (license 20) (modified by me to improve code and conform rest of
+ function to coding guidelines) ........
+
+2008-02-27 16:46 +0000 [r104592] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 104591 via svnmerge ........ r104591 |
+ russell | 2008-02-27 10:45:00 -0600 (Wed, 27 Feb 2008) | 4 lines
+ When we receive a known alarm, make sure that the unknown alarm
+ flag is not still set to make sure that when we come back out of
+ alarm, it gets reported in the log and manager interface (after
+ discussion with tzafrir on the -dev list) ........
+
+2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When queueing up a device state change when
+ the peer is loaded from the configuration give it a state of not
+ in use. We have to do this because the channel technology may not
+ yet be registered so the state could not be queried and would be
+ considered invalid. (closes issue #12087) Reported by: liorm
+
+ * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4
+ lines Only stop the MWI monitor thread if it was actually
+ started. (closes issue #12086) Reported by: francesco_r ........
+
+2008-02-27 15:34 +0000 [r104534] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is
+ specified. (Closes issue #12083)
+
+2008-02-27 15:31 +0000 [r104533] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression
+ introduced by state changes. (closes issue #12078) Reported by:
+ dimas Patches: v1-12078.patch uploaded by dimas (license 88)
+ (closes issue #12074) Reported by: Ivan
+
+2008-02-27 08:20 +0000 [r104502] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc, configs/vpb.conf.sample,
+ include/asterisk/module.h: Bring Voicetronix driver up to date
+ with current drivers (closes issue #12084) Reported by: mmickan
+ Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
+ module.h.diff uploaded by mmickan (license 400) vpb.conf.sample
+ uploaded by mmickan (license 400)
+
+2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant <russell@digium.com>
+
+ * doc/janitor-projects.txt: note that the chan_sip conversion is
+ already in progress
+
+ * doc/janitor-projects.txt: add another janitor project
+
+ * doc/janitor-projects.txt: Add the stuff from the janitor projects
+ page that is still relevant. I figure that if we keep this in the
+ tree, it will be much easier to keep up to date. The page on
+ asterisk.org just links to this on svn.digium.com/view
+
+2008-02-27 03:52 +0000 [r104418] Jason Parker <jparker@digium.com>
+
+ * doc/janitor-projects.txt (added): Create placeholder file...for
+ now.
+
+2008-02-27 02:05 +0000 [r104388] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Whitespace changes only
+
+2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008)
+ | 3 lines Avoid some recursion in the cleanup code for the
+ chanspy datastore (closes issue #12076, reported by junky,
+ patched by me) ........
+
+ * /: Blocked revisions 104332 via svnmerge ........ r104332 |
+ russell | 2008-02-26 18:54:29 -0600 (Tue, 26 Feb 2008) | 5 lines
+ Zaptel 1.4 now exposes FXO battery state as an alarm. However,
+ Asterisk 1.4 does not know what to do with these alarms. Only
+ Asterisk 1.6 cares about it. So, if we get an unknown alarm in
+ chan_zap, don't generate confusing log messages about it.
+ ........
+
+2008-02-26 22:14 +0000 [r104301] Steve Murphy <murf@digium.com>
+
+ * res/snmp/agent.c: small change to allow this file to compile. No
+ problem if you don't install the libsnmp package.
+
+2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: I swear I compiled this ... *cough*
+
+ * res/res_phoneprov.c: fix this module, too
+
+ * funcs/func_version.c: fix this module
+
+ * Makefile, include/asterisk, build_tools/make_version_h (added):
+ Re-add the automatically generated version.h, so that modules can
+ include for making build time decisions for cross asterisk
+ version compatibility
+
+ * main/manager.c, channels/chan_sip.c, include/asterisk/version.h
+ (removed), build_tools/make_version_c, res/res_agi.c,
+ main/http.c, include/asterisk/ast_version.h (added): Rename
+ version.h to ast_version.h. Next, I will be re-adding version.h
+ as an automatically generated file like it used to be. This still
+ needs to be there for modules that have to check it to compile
+ against multiple asterisk versions.
+
+2008-02-26 19:14 +0000 [r104215] Joshua Colp <jcolp@digium.com>
+
+ * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an
+ 'e' option to ResetCDR which re-enables a CDR that has been
+ disabled. (closes issue #11170) Reported by: kratzers Patches:
+ ResetCDR.1.diff uploaded by kratzers (license 307)
+
+2008-02-26 18:40 +0000 [r104176] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while,
+ even around single statements. 2) Revise the argument parsing
+ section, showing use of the standard macros. 3) Fix a typo.
+
+2008-02-26 18:27 +0000 [r104140-104142] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 104141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) |
+ 1 line Add badshell to .PHONY target (thanks Kevin) ........
+
+ * Makefile, /: Merged revisions 104139 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) |
+ 2 lines Since all shells aren't as awesome as bash, we have to
+ fail if somebody tries to use a literal "~" in DESTDIR. ........
+
+2008-02-26 16:51 +0000 [r104137] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting and doxygen while waiting on an
+ airport...
+
+2008-02-26 16:36 +0000 [r104133-104136] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 104135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) |
+ 5 lines Revert previous abspath change. ...abspath is new in GNU
+ make 3.81. I feel so...defeated. Must find new fix! ........
+
+ * /, sounds/Makefile: Merged revisions 104132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) |
+ 9 lines Fix a very bizarre issue we were seeing with our buildbot
+ when using a DESTDIR that wasn't an absolute path (such as
+ DESTDIR=~/asterisk-1.4). Apparently what was happening, was that
+ some of the targets were being expanded to the full path, so $@
+ ended up being /root/asterisk-1.4/[...]/ rather than
+ ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
+ in GNU make. (*cough*
+ http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
+ ........
+
+2008-02-26 14:51 +0000 [r104127] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Remove more hardcoded pipe symbols and replace
+ with commas. (closes issue #12072) Reported by: SimonSharman
+ Patches: features.patch uploaded by SimonSharman (license 410)
+ Tested by: SimonSharman
+
+2008-02-26 06:43 +0000 [r104125] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Use the readhandle for reads (closes issue
+ #12069)
+
+2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c: Add a \todo to convert this module to the event
+ system
+
+ * CHANGES: Update CHANGES for SMDI stuff
+
+ * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample,
+ include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions
+ 104119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008)
+ | 33 lines Merge changes from team/russell/smdi-1.4 This commit
+ brings in a significant set of changes to the SMDI support in
+ Asterisk. There were a number of bugs in the current
+ implementation, most notably being that it was very likely on
+ busy systems to pop off the wrong message from the SMDI message
+ queue. So, this set of changes fixes the issues discovered as
+ well as introducing some new ways to use the SMDI support which
+ are required to avoid the bugs with grabbing the wrong message
+ off of the queue. This code introduces a new interface to SMDI,
+ with two dialplan functions. First, you get an SMDI message in
+ the dialplan using SMDI_MSG_RETRIEVE() and then you access
+ details in the message using the SMDI_MSG() function. A side
+ benefit of this is that it now supports more than just chan_zap.
+ For example, with this implementation, you can have some FXO
+ lines being terminated on a SIP gateway, but the SMDI link in
+ Asterisk. Another issue with the current implementation is that
+ it is quite common that the station ID that comes in on the SMDI
+ link is not necessarily the same as the Asterisk voicemail box.
+ There are now additional directives in the smdi.conf
+ configuration file which let you map SMDI station IDs to Asterisk
+ voicemail boxes. Yet another issue with the current SMDI support
+ was related to MWI reporting over the SMDI link. The current code
+ could only report a MWI change when the change was made by
+ someone calling into voicemail. If the change was made by some
+ other entity (such as with IMAP storage, or with a web interface
+ of some kind), then the MWI change would never be sent. The SMDI
+ module can now poll for MWI changes if configured to do so. This
+ work was inspired by and primarily done for the University of
+ Pennsylvania. (also related to issue #9260) ........
+
+2008-02-26 00:05 +0000 [r104112] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 104111 via svnmerge ........ r104111 | qwell
+ | 2008-02-25 18:03:30 -0600 (Mon, 25 Feb 2008) | 7 lines
+ IPTOS_MINCOST is not defined on Solaris. (closes issue #12050)
+ Reported by: asgaroth Patches: 12050.patch uploaded by putnopvut
+ (license 60) ........
+
+2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option
+ in favor of dialplan substring variable syntax. (closes issue
+ #12060)
+
+ * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008)
+ | 10 lines This patch fixes some pretty significant problems with
+ how app_chanspy handles pointers to channels that are being spied
+ upon. It was very likely that a crash would occur if the channel
+ being spied upon hung up. This was because the current
+ ast_channel handling _requires_ that the object is locked or else
+ it could disappear at any time (except in the owning channel
+ thread). So, this patch uses some channel datastore magic on the
+ spied upon channel to be able to detect if and when the channel
+ goes away. (closes issue #11877) (patch written by me, but thanks
+ to kpfleming for the idea, and to file for review) ........
+
+ * /, main/utils.c: Merged revisions 104102 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008)
+ | 7 lines Improve the lock tracking code a bit so that a bunch of
+ old locks that threads failed to lock don't sit around in the
+ history. When a lock is first locked, this checks to see if the
+ last lock in the list was one that was failed to be locked. If it
+ is, then that was a lock that we're no longer sitting in a
+ trylock loop trying to lock, so just remove it. (inspired by
+ issue #11712) ........
+
+2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be
+ saved in Postgres. Note that these changes are
+ backward-compatible, so no changes to UPGRADE.txt are necessary.
+ (closes issue #9279) Reported by: rottenroddy Patches:
+ 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+ * funcs/func_global.c: Shared space for variables (instead of
+ letting other channels muck with your own) (closes issue #11943)
+ Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: jmls
+
+ * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25
+ Feb 2008) | 5 lines If the destination folder is full, don't
+ delete a message when exiting. (closes issue #12065) Reported by:
+ selsky Patch by: (myself) ........
+
+2008-02-25 21:40 +0000 [r104096] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6
+ lines Make it so a users.conf user creates both a SIP peer and a
+ SIP user. The user will be used for inbound authentication for
+ the device, and peer will be used for placing calls to the
+ device. (closes issue #9044) Reported by: queuetue Patches:
+ sip-gui-friend.diff uploaded by qwell (license 4) ........
+
+2008-02-25 20:50 +0000 [r104093] Jason Parker <jparker@digium.com>
+
+ * /, main/config.c: Merged revisions 104092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) |
+ 11 lines Allow the use of #include and #exec in situations where
+ the max include depth was only 1. Specifically, this fixes using
+ #include and #exec in extconfig.conf. This was basically caused
+ because the config file itself raises the include level to 1. I
+ opted not to raise the include limit, because recursion here
+ could cause very bizarre behavior. Pointed out, and tested by
+ jmls (closes issue #12064) ........
+
+2008-02-25 19:02 +0000 [r104089] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Instead of outputting a verbose message
+ every so often let's make it a debug message.
+
+2008-02-25 19:00 +0000 [r104088] Brett Bryant <bbryant@digium.com>
+
+ * doc/siptls.txt, configs/sip.conf.sample: Adding more tls
+ configuration details to sip.conf sample, with a list of valid
+ ciphers provided in both files. .. First commit since July, woot
+
+2008-02-25 18:38 +0000 [r104087] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25
+ Feb 2008) | 4 lines Ensure that the channel doesn't disappear in
+ agent_logoff(). If it does, it could cause a crash. (fixes the
+ crash reported in BE-396) ........
+
+2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6
+ lines If a resubscription comes in for a dialog we no longer know
+ about tell the remote side that the dialog does not exist so they
+ subscribe again using a new dialog. (closes issue #10727)
+ Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file
+ (license 11) ........
+
+ * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6
+ lines Due to recent changes tag will no longer be NULL if not
+ present so we have to use ast_strlen_zero to see if it's actually
+ blank. (closes issue #12061) Reported by: flefoll Patches:
+ chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll
+ (license 244) ........
+
+ * res/res_config_pgsql.c: Fix building of trunk. dbpass is always
+ going to exist.
+
+2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy <murf@digium.com>
+
+ * channels/chan_sip.c: Enforce a space between function args as per
+ code review.
+
+ * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned
+ on, and pgsql installed, I get warnings that stops the compile.
+ They are fixed now.
+
+2008-02-22 23:56 +0000 [r104045] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_zap.c, configure, configure.ac: Add protection to
+ chan_zap build when NEONMWI events are not defined
+
+2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES,
+ include/asterisk/manager.h: Move Originate to a separate
+ privilege and require the additional System privilege to call out
+ to a subshell.
+
+ * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008)
+ | 6 lines Backwards debug message. (closes issue #12052) Reported
+ by: flefoll Patches: chan_sip.c.br14.patch_found-notfound
+ uploaded by flefoll (license 244) ........
+
+ * res/res_config_pgsql.c: Allow database password to be NULL and
+ several other cleanups. (closes issue #12048) Reported by: bukaj
+ Patches: 20080222__bug12048.diff.txt uploaded by Corydon76
+ (license 14) Tested by: bukaj
+
+2008-02-21 21:27 +0000 [r104031] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix a typo
+
+2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 104027 via svnmerge ........ r104027 |
+ mmichelson | 2008-02-21 15:05:42 -0600 (Thu, 21 Feb 2008) | 4
+ lines And as a followup to revision 104026, completely remove
+ event-related calls from a section of code where we know there
+ was no event to handle or get. ........
+
+ * /: Blocked revisions 104026 via svnmerge ........ r104026 |
+ mmichelson | 2008-02-21 14:12:38 -0600 (Thu, 21 Feb 2008) | 7
+ lines Remove an incorrect debug message. It reported that it had
+ received a specific event and tried to report which event was
+ received. What actually was happening was that it was reporting
+ the number of bytes returned from a call to read(). Thanks to
+ Jared Smith for bringing the issue up on IRC ........
+
+ * res/res_agi.c: Instead of a notice, make the message about a
+ hung-up channel a debug message, and revert the original logic on
+ the if statement. Thanks to Juggie for bringing this to my
+ attention.
+
+2008-02-21 17:38 +0000 [r104024] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_zap.c: Added configuration distinction between neon
+ and fsk mwi detection Add the detection for neon MWI events got
+ rid of extraneous handle_init_event call in monitor loop
+
+2008-02-21 16:46 +0000 [r104020] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_agi.c: Don't print the fact that we are using dead mode
+ in AGI if called from the 'h' extension since it is well-known
+ that it will be running in dead mode. (closes issue #12046)
+ Reported by: explidous
+
+2008-02-21 16:44 +0000 [r104019] Joshua Colp <jcolp@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Disable epoll as it has caused more obscure issues then any of my
+ previous code. I will continue to work on it in a separate branch
+ to make it stable for a release and test it against the following
+ issues. (closes issue #11253) Reported by: falves11 (closes issue
+ #11657) Reported by: davevg (closes issue #11033) Reported by:
+ falves11
+
+2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c, /: Merged revisions 104015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb
+ 2008) | 2 lines reduce the likelihood that HTTP Manager session
+ ids will consist of primarily '1' bits ........
+
+2008-02-21 05:21 +0000 [r104014] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astman.c: Ignore some more unused generated events. (closes
+ issue #12042) Reported by: junky Patches: astman_events.diff
+ uploaded by junky (license 177)
+
+2008-02-20 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-beta4 released.
+
+2008-02-20 22:34 +0000 [r103957] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb
+ 2008) | 8 lines Clear up confusion when viewing the
+ QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
+ user's perspective, the queue does exist, we shouldn't tell them
+ we couldn't find the queue. Instead since it is a dead queue,
+ report a 0 waiting count This issue was brought up on IRC by jmls
+ ........
+
+2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_h323.c: Try to do Packet2Packet bridging with
+ chan_h323 if reinviting isn't enabled. (closes issue #11901)
+ Reported by: pj
+
+ * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6
+ lines Don't wait for additional digits when overlap dialing is
+ enabled if the setup message contains the sending_complete
+ information element. (closes issue #11785) Reported by: klaus3000
+ Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt
+ uploaded by klaus3000 (license 65) ........
+
+2008-02-20 21:41 +0000 [r103908] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 103904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed,
+ 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL
+ while attempting to lock it. (closes issue #12039) Reported by:
+ danpwi ........
+
+2008-02-20 21:36 +0000 [r103903] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone
+ detection routines. Separate fax detection from digit detected.
+ Added CED (called) tone detection for fax (previously, only CNG
+ (calling) was supported). Separate DTMF/MF code paths where
+ appropriate. Allow detection of arbitary tones. (closes issue
+ #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch
+ uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache
+
+2008-02-20 21:08 +0000 [r103902] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix a crash due to the wrong variable being
+ used when building a directory string. (closes issue #12027)
+ Reported by: jaroth Patches: forward.patch uploaded by jaroth
+ (license 50) Tested by: jaroth
+
+2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/sched.h: Add some documentation fixups
+
+ * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20
+ Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022)
+ Reported by: asgaroth Patches: 20080219__bug12022.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: asgaroth ........
+
+2008-02-20 15:21 +0000 [r103844] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't
+ been converted to ',' (closes issue #12034) Reported by: kowalma
+
+2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c: *mumble*
+
+ * main/audiohook.c: file not found.
+
+ * main/audiohook.c: Minor test...
+
+2008-02-20 00:49 +0000 [r103833] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: When using IMAP storage, if the folder you
+ attempt to save to does not exist, create it first. (closes issue
+ #12032) Reported by: jaroth Patches: createfolder.patch uploaded
+ by jaroth (license 50) Tested by: jaroth
+
+2008-02-19 22:35 +0000 [r103831-103832] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Make sure to mask out non-audio first as well
+
+ * main/channel.c: Maybe we should set the value before we test it?
+ Fixes an issue people have been seeing (unreported?) with file
+ playback not working.
+
+2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp <jcolp@digium.com>
+
+ * main/loader.c: Add a log message that appears when you try to
+ unload a module that isn't loaded. (closes issue #12033) Reported
+ by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by
+ jamesgolovich (license 176)
+
+ * main/file.c: Only output a log message saying the format does not
+ exist if it actually does not exist, not if the file itself could
+ not be opened. (closes issue #11828) Reported by: IgorG Patches:
+ readfile.v1.diff uploaded by IgorG (license 20)
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6
+ lines Send CallerID Name in setup message. (closes issue #11241)
+ Reported by: tusar Patches: h323id_as_callerid_name.patch
+ uploaded by tusar (license 344) ........
+
+2008-02-19 20:06 +0000 [r103822] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 103821 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19
+ Feb 2008) | 8 lines Account for the fact that the "other" channel
+ can disappear while the local pvt is not locked. (fixes a problem
+ introduced in rev 100581) (closes issue #12012) Reported by:
+ stevedavies Patch by me ........
+
+2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_authenticate.c: len already contains the position we
+ want to examine, if we move one left again we'll actually
+ probably be looking at a digit. (issue #12030) Reported by:
+ alligosh
+
+ * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add
+ CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan
+ application. This will either be set to NOCHANNEL if the given
+ channel was not found or SUCCESS if it worked. (closes issue
+ #11553) Reported by: johan Patches:
+ UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
+ CHANGES.channelredirect.patch uploaded by johan (license 334)
+ app_channelredirect-20080219.patch uploaded by johan (license
+ 334)
+
+2008-02-19 18:14 +0000 [r103818] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm
+ Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm
+ (license 99) Patch fixes problem of device state incorrectly
+ reporting idle before PBX answers incoming call on FXO channel.
+ Device status is updated now during new channel creation.
+
+2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp <jcolp@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 103812 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb
+ 2008) | 4 lines Don't look for launchd when cross compiling.
+ (closes issue #12029) Reported by: ovi ........
+
+ * /: Blocked revisions 103807 via svnmerge ........ r103807 | file
+ | 2008-02-19 11:01:42 -0400 (Tue, 19 Feb 2008) | 2 lines Fix
+ building of chan_sip. ........
+
+2008-02-19 00:59 +0000 [r103805] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Change verbosity into debug for Hebrew (and various
+ whitespace fixes) (Closes issue #12011)
+
+2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 103801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) |
+ 10 lines Ensure that emulated DTMFs do not get interrupted by
+ another begin frame. (closes issue #11740) Reported by: gserra
+ Patches: v1-11740.patch uploaded by dimas (license 88) (closes
+ issue #11955) Reported by: tsearle (closes issue #10530) Reported
+ by: xmarksthespot ........
+
+ * main/channel.c, main/frame.c, channels/chan_sip.c,
+ include/asterisk/channel.h, include/asterisk/frame.h: Add a
+ non-invasive API for application level manipulation of T38 on a
+ channel. This uses control frames (so they can even pass across
+ IAX2) to negotiate T38 and provided a way of getting the current
+ status of T38 using queryoption. This should by no means cause
+ any issues and if it does I will take responsibility for it.
+ (closes issue #11873) Reported by: dimas Patches:
+ v4-t38-api.patch uploaded by dimas (license 88)
+
+ * main/frame.c: Add some missing control frames.
+
+2008-02-18 22:33 +0000 [r103796] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) |
+ 1 line Fix previous commit so that we actually disable
+ echocanbridged if echocancel is off. ........
+
+2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Commit chan_zap portion of #11964: add the
+ ability to get ORIG_CALLED_NUM
+
+2008-02-18 21:30 +0000 [r103791] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) |
+ 4 lines Correct a message when echocancelwhenbridged is on, but
+ echocancel is not. Closes issue #12019 ........
+
+2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes
+ in. Also add support for multiple DPCs per linkset. #11779
+
+2008-02-18 20:53 +0000 [r103787] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/app.c: Merged revisions 103786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb
+ 2008) | 10 lines There was an invalid assumption when calculating
+ the duration of a file that the filestream in question was
+ created properly. Unfortunately this led to a segfault in the
+ situation where an unknown format was specified in voicemail.conf
+ and a voicemail was recorded. Now, we first check to be sure that
+ the stream was written correctly or else assume a zero duration.
+ (closes issue #12021) Reported by: jakep Tested by: putnopvut
+ ........
+
+2008-02-18 19:47 +0000 [r103783] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: make the output of 'core show settings' a bit
+ nicer. (closes issue #12020) Reported by: seanbright Patches:
+ asterisk.c.patch uploaded by seanbright (license 71)
+
+2008-02-18 17:45 +0000 [r103781] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008)
+ | 9 lines When a SIP channel is being auto-destroyed, it's
+ possible for it to still be in bridge code. When that happens, we
+ crash. Delay the RTP destruction until the bridge is ended.
+ (closes issue #11960) Reported by: norman Patches:
+ 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: norman ........
+
+2008-02-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-beta3 released.
+
+2008-02-18 17:12 +0000 [r103772] Olle Johansson <oej@edvina.net>
+
+ * main/channel.c, channels/chan_sip.c: Make sure we can set up
+ calls without audio (text+video). And ... it works!
+
+2008-02-18 16:40 +0000 [r103771] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb
+ 2008) | 10 lines Fix a linked list corruption that under the
+ right circumstances could lead to a looped list, meaning it will
+ traverse forever. (closes issue #11818) Reported by: michael-fig
+ Patches: 11818.patch uploaded by putnopvut (license 60) Tested
+ by: michael-fig ........
+
+2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 103768 via svnmerge ........ r103768 | file
+ | 2008-02-18 12:11:51 -0400 (Mon, 18 Feb 2008) | 4 lines Backport
+ fix from issue #9325. (closes issue #11980) Reported by: rbrunka
+ ........
+
+ * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h:
+ Add an API call (ast_async_parseable_goto) which parses a goto
+ string and does an async goto instead of an explicit goto.
+ (closes issue #11753) Reported by: johan
+
+ * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2
+ lines Don't care if the extension given doesn't exist for
+ subscription based MWI. ........
+
+2008-02-18 10:10 +0000 [r103755] Olle Johansson <oej@edvina.net>
+
+ * CHANGES, channels/chan_iax2.c: - No space in manager event names,
+ please - Add new event to CHANGES
+
+2008-02-18 04:43 +0000 [r103754] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, main/channel.c, main/pbx.c,
+ funcs/func_channel.c, include/asterisk/channel.h, CHANGES,
+ main/cli.c: Context tracing for channels (closes issue #11268)
+ Reported by: moy Patches:
+ chantrace-datastored-encapsulated-rev94934.patch uploaded by moy
+ (license 222)
+
+2008-02-16 21:22 +0000 [r103750] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now
+ monitoring chan_skinny port with nagios or zabbix wont generate
+ noise on the console. @ok tilghman
+
+2008-02-15 23:32 +0000 [r103742] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15
+ Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race
+ condition (closes issue #11780) Reported by: guillecabeza
+ Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
+ 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
+ 380) ........
+
+2008-02-15 23:20 +0000 [r103740] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES: Document GotoIfTime change from svn revision 103738
+
+2008-02-15 23:14 +0000 [r103739] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related
+ to the use of AES encryption. 1024 was used instead of 128 when
+ using AES from OpenSSL. Many thanks to d1mas for figuring this
+ one out! (closes issue #11946) Reported by: bbhoss Patches:
+ v1-11946.patch uploaded by dimas (license 88)
+
+2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Add proper "false" case behavior to GotoIfTime
+ (closes issue #11719) Reported by: kshumard Patches:
+ gotoiftime.twobranches.patch uploaded by kshumard (license 92)
+ Tested by: kshumard
+
+ * apps/app_voicemail.c: Fix redeclaration of variables when using
+ IMAP storage (closes issue #11988) Reported by: jaroth Patches:
+ variable_cleanup.patch uploaded by jaroth (license 50)
+
+2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant <russell@digium.com>
+
+ * /, main/loader.c: Merged revisions 103728 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008)
+ | 4 lines In the case that you try to directly reload a module
+ has returned AST_MODULE_LOAD_DECLINE, log a message indicating
+ that the module is not fully initialized and must be initialized
+ using "module load". ........
+
+ * /, main/loader.c: Merged revisions 103726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008)
+ | 6 lines Don't attempt to execute the reload callback for a
+ module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
+ that was reported against chan_console in trunk. (closes issue
+ #11953, reported by junky, fixed by me) ........
+
+2008-02-15 17:32 +0000 [r103725] Mark Michelson <mmichelson@digium.com>
+
+ * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged
+ revisions 103722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb
+ 2008) | 8 lines Final round of changes for configure script logic
+ for IMAP Now if a directory is specified, then we will search
+ that directory for a source installation of the IMAP toolkit. If
+ none is found, then we will use that directory as the basis for
+ detecting a package installation of the IMAP c-client. If that
+ check fails, then configure will fail. ........
+
+2008-02-15 17:29 +0000 [r103723] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c,
+ include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c,
+ apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c,
+ main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h,
+ main/ast_expr2f.c, channels/misdn/ie.c,
+ channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c,
+ include/asterisk/strings.h, res/res_config_ldap.c,
+ include/asterisk/time.h: Fix up some doxygen issues. (closes
+ issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy
+ (license 35)
+
+2008-02-15 15:45 +0000 [r103716] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/conf2ael.c: Remove extraneous copy (closes issue #12002)
+ Reported by: junky Patches: conf2ael.diff uploaded by junky
+ (license 177)
+
+2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson <mmichelson@digium.com>
+
+ * configure, configure.ac: Merging of changes from 1.4 revision
+ 103713.
+
+ * /: Blocked revisions 103713 via svnmerge ........ r103713 |
+ mmichelson | 2008-02-15 09:05:49 -0600 (Fri, 15 Feb 2008) | 8
+ lines Fix a bit of wrong logic in the configure script that
+ caused problems when trying to configure without IMAP. Patch
+ suggestion from phsultan, but I modified it slightly. (closes
+ issue #12003) Reported by: pj Tested by: putnopvut ........
+
+ * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as
+ made to 1.4 in revision 103710
+
+ * /: Blocked revisions 103709 via svnmerge ........ r103709 |
+ mmichelson | 2008-02-14 18:50:49 -0600 (Thu, 14 Feb 2008) | 6
+ lines I apparently misunderstood one of the requirements of this
+ configure change. Now, if a source directory is specified with
+ the --with-imap option, and a valid source installation is not
+ detected there, then configure will fail and will not check for a
+ package installation. ........
+
+ * doc/tex/imapstorage.tex: Trunk version of 1.4's imap
+ documentation updates
+
+ * /: Blocked revisions 103703 via svnmerge ........ r103703 |
+ mmichelson | 2008-02-14 17:49:24 -0600 (Thu, 14 Feb 2008) | 3
+ lines Make a small clarification in the documentation ........
+
+ * /: Blocked revisions 103701 via svnmerge ........ r103701 |
+ mmichelson | 2008-02-14 17:44:17 -0600 (Thu, 14 Feb 2008) | 3
+ lines Update documentation regarding configuration of IMAP
+ ........
+
+ * configure, configure.ac: See commit message for svn revision
+ 103698. This behavior is the same as what is described there. The
+ difference is that trunk already had the --with-imap=system
+ option, but it only checked the include path for headers in the
+ imap directory and not also the c-client directory.
+
+ * /: Blocked revisions 103698 via svnmerge ........ r103698 |
+ mmichelson | 2008-02-14 17:30:17 -0600 (Thu, 14 Feb 2008) | 13
+ lines Change to the configure logic regarding IMAP. Prior to this
+ commit, if you wished to configure Asterisk with IMAP support,
+ you would use the --with-imap configure switch in one of the
+ following two ways: --with-imap=/some/directory would look in the
+ directory specified for a UW IMAP source installation --with-imap
+ would assume that you had imap-2004g installed in .. relative to
+ the Asterisk source With this set of changes the two above
+ options still work the same, but there are two new behaviors,
+ too. --with-imap=system will assume that you have -libc-client.so
+ where you store your shared objects and will attempt to find
+ c-client headers in your include path either in the imap or
+ c-client directory. If either of the two original methods of
+ specifying the imap option should fail, then the check for
+ --with-imap =system will be performed in addition. It is only
+ after this "system" check that failure can happen. ........
+
+2008-02-14 21:21 +0000 [r103694] Jason Parker <jparker@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify
+ ldap autoconf function, so that an incorrect ldap library is not
+ found on Solaris (it is incompatible). Also removes second check
+ for awk, which causes Solaris to find an incompatible version of
+ awk. (closes issue #11829) Reported by: snuffy Patches:
+ bug-11829.diff uploaded by snuffy (license 35)
+
+2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu,
+ 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........
+
+ * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu,
+ 14 Feb 2008) | 9 lines Fix the new message count if delete=yes
+ when using IMAP storage. (closes issue #11406) Reported by:
+ jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
+ 50) Tested by: jaroth ........
+
+ * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change
+ the queue holdtime announcement to happen at any interval (not
+ just greater than two minutes). Remove the saying of less-than
+ for holdtime announcements since it can lead to awkward holdtime
+ announcements. Using '1' as a queue-round-seconds value is no
+ longer valid. (closes issue #9736) Reported by: caio1982 Patches:
+ queue_announce5.diff uploaded by caio1982 (license 22) Tested by:
+ caio1982, putnopvut
+
+2008-02-14 19:52 +0000 [r103685] Jason Parker <jparker@digium.com>
+
+ * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) |
+ 5 lines Document the 'l' option to the CDR() function. (Thanks
+ voipgate for pointing out the option, and Leif for providing text
+ for it.) Closes issue #11695. ........
+
+2008-02-14 19:47 +0000 [r103682] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_externalivr.c: a few syntax changes and safer code
+
+2008-02-14 18:39 +0000 [r103677] Jason Parker <jparker@digium.com>
+
+ * channels/chan_iax2.c: Add periodic jitter stats to CLI and
+ manager. (closes issue #8188) Reported by: stevedavies Patches:
+ jblogging-trunk.patch uploaded by stevedavies
+ jblogging-trunk_wmgrevent.patch uploaded by johann8384
+ updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384
+ (license 190) (with additional changes by me) Tested by:
+ stevedavies, johann8384
+
+2008-02-14 10:19 +0000 [r103668] Olle Johansson <oej@edvina.net>
+
+ * res/res_agi.c, apps/app_externalivr.c: Formatting fixes
+
+2008-02-13 21:04 +0000 [r103662] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_externalivr.c: (closes issue #11825) Reported by:
+ ctooley Patches: additional_eivr_commands.patch uploaded by
+ ctooley (license 136) Tested by: ctooley
+
+2008-02-13 15:47 +0000 [r103658] Mark Michelson <mmichelson@digium.com>
+
+ * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold
+ and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold
+ (closes issue #11904) Reported by: dimas Patches: v2-moh.patch
+ uploaded by dimas (license 88) Tested by: dimas
+
+2008-02-13 06:35 +0000 [r103608] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 103607 via svnmerge ........ r103607 |
+ tilghman | 2008-02-13 00:25:03 -0600 (Wed, 13 Feb 2008) | 7 lines
+ We aren't talking to ourselves; we're talking to someone else.
+ (closes issue #11771) Reported by: msetim Patches:
+ ami_agent_talkingto-1.4.diff uploaded by caio1982 (license 22)
+ Tested by: caio1982, msetim ........
+
+2008-02-13 00:55 +0000 [r103559] Mark Michelson <mmichelson@digium.com>
+
+ * main/event.c: Fix a small logic error in ast_event_iterator_next.
+ The previous logic allowed for the iterator to indicate there was
+ more data than there really was, causing the iterator read beyond
+ the end of the event structure. This led to invalid memory reads
+ and potential crashes.
+
+2008-02-13 00:31 +0000 [r103557] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 103556 via svnmerge ........ r103556 |
+ tilghman | 2008-02-12 18:26:57 -0600 (Tue, 12 Feb 2008) | 7 lines
+ Refuse to load app_voicemail if res_adsi is not loaded (which is
+ a symbol dependency) (closes issue #11760) Reported by:
+ non-poster Patches: 20080114__bug11760.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: Corydon76, non-poster,
+ jamesgolovich ........
+
+2008-02-12 22:26 +0000 [r103447-103506] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 103503 via svnmerge ........ r103503 | qwell
+ | 2008-02-12 16:22:54 -0600 (Tue, 12 Feb 2008) | 1 line Remove
+ condition that was impossible. ........
+
+ * main/manager.c: Even more sane permissions. This should be
+ handled via a umask, like in many other places.
+
+ * main/manager.c: Use slight more sane permissions
+
+2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Remove development version notice.
+
+ * main/manager.c: Fix build on *BSD. These permissions constants
+ are not available there.
+
+2008-02-12 15:13 +0000 [r103386] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4
+ lines Even if no CallerID name or number has been provided by the
+ remote party still use the configured sip.conf ones. (closes
+ issue #11977) Reported by: pj ........
+
+2008-02-12 14:08 +0000 [r103341] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags
+ structure in aji_client and aji_buddy rather than an integer.
+ Modify calls to various ast_*_flag macros accordingly.
+
+2008-02-12 00:24 +0000 [r103331] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, include/asterisk/config.h, CHANGES,
+ main/config.c: Requested changes from Pari, reviewed by Russell.
+ Added ability to retrieve list of categories in a config file.
+ Added ability to retrieve the content of a particular category.
+ Added ability to empty a context. Created new action to create a
+ new file. Updated delete action to allow deletion by line number
+ with respect to category. Added new action insert to add new
+ variable to category at specified line. Updated action newcat to
+ allow new category to be inserted in file above another existing
+ category.
+
+2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4
+ lines If entering a conference with the 'w' option ensure that we
+ can't listen or speak until the marked user appears. (closes
+ issue #11835) Reported by: alanmcmillan ........
+
+ * res/res_agi.c: Remove ast_module_user usage from res_agi. This is
+ taken care of in the core.
+
+ * main/pbx.c, main/manager.c, main/translate.c, main/logger.c,
+ main/app.c, main/utils.c, main/indications.c, main/asterisk.c,
+ main/rtp.c: Just some minor coding style cleanup...
+
+ * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue
+ #10661) Reported by: junky
+
+2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/zapata.conf.sample: Merged revisions 103315 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb
+ 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared)
+ ........
+
+2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, channels/chan_iax2.c: Add support for allowing a
+ native bridge to happen when the L option is enabled. The RTP
+ bridging could already handle this, it just needed to be enabled
+ in the main bridging code. (issue #10647) Reported by: samdell3
+
+ * channels/chan_skinny.c: Change chan_skinny to use debug messages
+ as appropriate. (closes issue #11967) Reported by: mvanbaak
+ Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak
+ (license 7)
+
+2008-02-11 06:05 +0000 [r103306] James Golovich <james@gnuinter.net>
+
+ * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip
+ on reload (issue #11930)
+
+2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix improper indentation. Thanks again to
+ snuffy for pointing it out.
+
+ * apps/app_queue.c: Add a couple of comments to clarify the
+ unreffing of queues. Thanks to snuffy for the idea.
+
+ * main/event.c: Fix a problem regarding network vs. host byte order
+ in the event API. ast_event_iterator_get_ie_type should return
+ the ie type in host byte order. Furthermore, ast_event_get_ie_raw
+ should already have its ie type argument in host byte order since
+ it could be called externally (and it in fact is called in this
+ way by ast_event_get_cached).
+
+2008-02-09 11:27 +0000 [r103249] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c,
+ apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c,
+ apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c,
+ apps/app_channelredirect.c, apps/app_directed_pickup.c,
+ apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c,
+ apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c:
+ whitespace fixes only.
+
+2008-02-09 06:33 +0000 [r103198] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09
+ Feb 2008) | 4 lines Commit fix for being unable to send voicemail
+ from VoiceMailMain Reported by: William F Acker (via the -users
+ mailing list) Patch by: Corydon76 (license 14) ........
+
+2008-02-08 21:26 +0000 [r103171] Russell Bryant <russell@digium.com>
+
+ * main/udptl.c, main/pbx.c, channels/chan_sip.c,
+ channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c,
+ main/rtp.c, channels/chan_usbradio.c, main/cdr.c,
+ channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c,
+ pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c,
+ apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c,
+ apps/app_voicemail.c: Merge changes from
+ team/mvanbaak/cli-command-audit (closes issue #8925) About a year
+ ago, as Leif Madsen and Jim van Meggelen were going over the CLI
+ commands in Asterisk 1.4 for the next version of their book, they
+ documented a lot of inconsistencies. This set of changes
+ addresses all of these issues and has been reviewed by Leif.
+ While this does introduce even more changes to the CLI command
+ structure, it makes everything consistent, which is the most
+ important thing. Thanks to all that helped with this one!
+
+2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes
+ different arguments in trunk than 1.4.
+
+ * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb
+ 2008) | 10 lines Prevent a potential three-thread deadlock. Also
+ added a comment block to explicitly state the locking order
+ necessary inside app_queue. (closes issue #11862) Reported by:
+ flujan Patches: 11862.patch uploaded by putnopvut (license 60)
+ Tested by: flujan ........
+
+ * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri,
+ 08 Feb 2008) | 6 lines Yield the thread and return -1 if the
+ ioctl fails for Zaptel timing device. (closes issue #11891)
+ Reported by: tzafrir ........
+
+2008-02-08 16:49 +0000 [r103044] Russell Bryant <russell@digium.com>
+
+ * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At
+ the request of ManxPower, include the UPGRADE.txt from 1.2 and
+ 1.4, as well. This way, if people need to go back and review what
+ was deprecated in previous major releases, it is readily
+ available to them. Thanks for the suggestion!
+
+2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a network byte order issue and ensure
+ when creating an outgoing dialog that the socket always contains
+ information such as type and port. (closes issue #11916) Reported
+ by: mnnojd
+
+ * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb
+ 2008) | 4 lines Make sure the presence of dbsecret is factored
+ into user scoring. (closes issue #11952) Reported by: bbhoss
+ ........
+
+2008-02-07 21:37 +0000 [r102933] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: This is a combination new feature/bug fix for
+ app_chanspy. New feature: Add the 'e' option, which takes as an
+ argument a list of interfaces separated by colons. This way, you
+ will only be able to spy on this limited list of interfaces. Bug
+ fix: change some pointer checks to ast_strlen_zero so that spying
+ would work properly even if no channel was specified as the first
+ argument to chanspy. (closes issue #10072) Reported by:
+ xmarksthespot Patches:
+ bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by
+ xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak
+
+2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_adsiprog.c: whitespace fixes only
+
+ * apps/app_alarmreceiver.c: There she goes! First commit from me to
+ trunk \o/ Make app_alarmreceiver honor code guidelines and fix
+ whitespace errors. No functional changes.
+
+2008-02-07 20:02 +0000 [r102859] Jason Parker <jparker@digium.com>
+
+ * /, main/features.c: Merged revisions 102858 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) |
+ 7 lines Specify which digit string was matched in debug message.
+ (closes issue #11949) Reported by: dimas Patches:
+ v1-feature-debug.patch uploaded by dimas (license 88) ........
+
+2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/zapata.conf.sample: Merged revisions 102807 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb
+ 2008) | 2 lines document usage of 'transfer' configuration option
+ for ISDN PRI switch-side transfers ........
+
+2008-02-06 20:12 +0000 [r102777] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add the channel's unique id to the AgentCalled
+ manager event to make it more consistent with other manager
+ events.
+
+2008-02-06 18:01 +0000 [r102726] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2
+ lines Only consider a T.38-only INVITE compatible if we have both
+ a joint capability between us and them and if they provided T.38.
+ ........
+
+2008-02-06 16:23 +0000 [r102700] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY
+ dialplan functions provided by sergee. I just added the ability
+ to set multiple fields at once after discussions with Tilghman
+ and Russell. Currently limited to 30 fields. (closes issue
+ #11887) Reported by: sergee Patches:
+ rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy
+ (license 396) Tested by: sergee, otherwiseguy
+
+2008-02-06 15:46 +0000 [r102654] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 102653 via svnmerge ........ r102653 | file
+ | 2008-02-06 11:43:38 -0400 (Wed, 06 Feb 2008) | 4 lines Add
+ missing header file and ASTERISK_FILE_VERSION usage. (closes
+ issue #11936) Reported by: snuffy ........
+
+2008-02-06 15:20 +0000 [r102652] Russell Bryant <russell@digium.com>
+
+ * /, configs/features.conf.sample: Merged revisions 102651 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008)
+ | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets
+ inherited by outbound channels. (due to a discussion between me
+ and a user via email) ........
+
+2008-02-06 03:05 +0000 [r102602] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05
+ Feb 2008) | 4 lines Move around some defines to unbreak ODBC
+ storage. (closes issue #11932) Reported by: snuffy ........
+
+2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Remove an extra debug message I left in
+
+ * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
+ apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c,
+ apps/app_followme.c, main/loader.c, channels/chan_usbradio.c,
+ main/tcptls.c, res/res_agi.c, apps/app_minivm.c,
+ apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c,
+ main/astmm.c: Get rid of any remaining ast_verbose calls in the
+ code in favor of ast_verb (closes issue #11934) Reported by:
+ mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by
+ mvanbaak (license 7)
+
+ * apps/app_voicemail.c: Change verbose messages to use the ast_verb
+ macro. (closes issue #11931) Reported by: snuffy Patches:
+ bug-11931.diff uploaded by snuffy (license 35)
+
+2008-02-05 20:51 +0000 [r102500] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c: Change where priority of a goto is adjusted.
+ Partially reverts 102272. Closes issue #11929 (credit to file for
+ fix suggestion - we still <3 you)
+
+2008-02-05 20:03 +0000 [r102454] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue,
+ 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes
+ issue #11919) Reported by: eferro Patches:
+ mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
+ Tested by: eferro ........
+
+2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot
+ me.
+
+ * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3
+ lines If a REGISTER attempt comes in that is a retransmission of
+ a previous REGISTER do not create a new nonce value. (issue
+ #BE-381) ........
+
+ * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb
+ 2008) | 4 lines Perform dialing asynchronously when using the
+ originate CLI command so the CLI does not appear to block.
+ (closes issue #11927) Reported by: bbhoss ........
+
+2008-02-04 21:15 +0000 [r102329] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, main/asterisk.c: Merged revisions 102323 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008)
+ | 7 lines Cross-platform fix: OS X now deprecates the use of the
+ daemon(3) API. (closes issue #11908) Reported by: oej Patches:
+ 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76 ........
+
+2008-02-04 18:39 +0000 [r102297] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Add line numbers to warning/error messages
+ (and pretty up some existing ones). (closes issue #11894)
+ Reported by: jmls Patches: chan_zap.patch uploaded by jmls
+ (license 141)
+
+2008-02-04 15:16 +0000 [r102272] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Update handling of asyncgoto so it properly works on
+ channels that are currently executing a PBX. (closes issue
+ #11914) Reported by: arnd (closes issue #11753) Reported by:
+ johan
+
+2008-02-04 14:37 +0000 [r102262] Jason Parker <jparker@digium.com>
+
+ * configs/extensions.ael.sample, configs/extensions.lua.sample:
+ Change examples to use G here also. Closes issue #11875
+
+2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03
+ Feb 2008) | 6 lines Missing braces. (closes issue #11912)
+ Reported by: dimas Patches: sprintf.patch uploaded by dimas
+ (license 88) ........
+
+ * main/manager.c: CoreSettings and CoreStatus are missing the
+ terminating "\r\n". Also, some miscellaneous spacing and
+ initialization issues. (closes issue #11909) Reported by: srt
+ Patches: patch-11909-2.diff uploaded by srt (license 378) Tested
+ by: srt
+
+2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8
+ lines Use the same CSEQ on CANCEL as on INVITE (according to RFC
+ 3261) (closes issue #9492) Reported by: kryptolus Patches:
+ bug9492.txt uploaded by oej (license 306) Tested by: oej ........
+
+ * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8
+ lines Handle ACK and CANCEL in an invite transaction - even if we
+ get INFO transactions during the actual call setup. (closes issue
+ #10567) Reported by: jacksch Tested by: oej Patch by: oej
+ inspired by suggestions from neutrino88 in the bug tracker
+ ........
+
+2008-02-03 06:43 +0000 [r102064] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Change the version number in the
+ configure script from 1.4 to 1.6
+
+2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h: The documentation page has to be in its
+ own comment block to work, apparently. Fix it up!
+
+ * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008)
+ | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that
+ even though G.722 is 16 kHz, it is supposed to specified as 8 kHz
+ in the RTP, and RTP timestamps are supposed to be calculated
+ based on 8 kHz. (Apparently this is due to a bug in a spec, but
+ people follow it anyway, because it's the spec ...) ........
+
+2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01
+ Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded
+ voicemail, and fixed several other bugs while I'm in the area.
+ (closes issue #11615) Reported by: jamessan Patches:
+ 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, jamessan ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 101894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008)
+ | 2 lines Change detection of getifaddrs to use
+ AST_C_COMPILE_CHECK, backported from trunk (as suggested by
+ kpfleming) ........
+
+ * res/res_config_curl.c: Fix multi, when using the LIKE query.
+ (closes issue #11889) Reported by: jmls Patches:
+ res_config_curl.patch uploaded by jmls (license 141) Tested by:
+ jmls
+
+2008-02-01 18:24 +0000 [r101869] Jason Parker <jparker@digium.com>
+
+ * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak.
+
+2008-02-01 18:08 +0000 [r101824] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling
+ functionality by changing the config file keyword
+
+2008-02-01 17:44 +0000 [r101823] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_authenticate.c: Move an feof() call to before the
+ fgets(). This would have exited the loop early if you had an
+ authentication file with no newline at the end.
+
+2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 101820 via svnmerge ........ r101820 |
+ russell | 2008-02-01 11:27:02 -0600 (Fri, 01 Feb 2008) | 1 line
+ off by one error ........
+
+ * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01
+ Feb 2008) | 4 lines Don't overwrite the last character of a line
+ if it's not a newline. This would happen if the last line in the
+ file doesn't have a newline. (pointed out by Qwell) ........
+
+2008-02-01 16:01 +0000 [r101773] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/acl.c: Merged revisions 101772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008)
+ | 2 lines Compatibility fix for OpenWRT (reported by Brian
+ Capouch via the mailing list) ........
+
+2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant <russell@digium.com>
+
+ * apps/app_authenticate.c: simplify some code, tweak formatting,
+ and reduce indentation
+
+ * apps/app_authenticate.c: reduce a level of indentation
+
+ * apps/app_channelredirect.c: Get rid of a goto where there was no
+ extra cleanup happening at the exit point
+
+ * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31
+ Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial
+ strings for the case that no peer or hostname was provided, which
+ is the one part of the dial string that is absolutely required.
+ If it's not there, bail out. (closes issue #11897) Reported by
+ sokhapkin Patch by me ........
+
+2008-02-01 00:08 +0000 [r101650] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan
+ 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle
+ periods of no channel activity" (closes issue #9256) Reported by:
+ cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
+ uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
+ rjain ........
+
+2008-01-31 23:14 +0000 [r101611] Russell Bryant <russell@digium.com>
+
+ * /, main/translate.c, main/file.c: Merged revisions 101601 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008)
+ | 12 lines Fix a couple of places where ast_frfree() was not
+ called on a frame that came from a translator. This showed itself
+ by g729 decoders not getting released. Since the flag inside the
+ translator frame never got unset by freeing the frame to indicate
+ it was no longer in use, the translators never got destroyed, and
+ thus the g729 licenses were not released. (closes issue #11892)
+ Reported by: xrg Patches: 11892.diff uploaded by russell (license
+ 2) Tested by: xrg, russell ........
+
+2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Forgot an !
+
+ * apps/app_queue.c: A change I made to accommodate the "linear"
+ strategy in trunk caused queue strategies to not be loaded from
+ realtime queues. This commit fixes that. Thanks to jmls for
+ pointing this problem out to me on IRC. This also contains some
+ changes to S_OR where it should be used. Thanks to Qwell for
+ pointing these out.
+
+2008-01-31 21:33 +0000 [r101577] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a simple deadlock that was introduced
+ _right_ before this code got merged into trunk. (closes issue
+ #11895, reported by pj, patched by me)
+
+2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Handle the case of a NULL state_interface when
+ checking a realtime member. Thanks to jmls for finding this
+ issue.
+
+ * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan
+ 2008) | 10 lines 1. Prevent the addition of an extra '/' to the
+ beginning of an absolute pathname. 2. If ast_monitor_change_fname
+ is called and the new filename is the same as the old, then exit
+ early and don't set the filename_changed field in the monitor
+ structure. Setting it in this case was causing ast_monitor_stop
+ to erroneously delete them. (closes issue #11741) Reported by:
+ garlew Tested by: putnopvut ........
+
+2008-01-31 19:54 +0000 [r101483] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 101482 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) |
+ 4 lines Solaris compat fixes for struct in_addr funkiness. Issue
+ #11885, patch by snuffy. ........
+
+2008-01-31 19:43 +0000 [r101481] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 101480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1
+ line closes issue #11845; that's the one where there's a 1004
+ byte cdr leak with every AMI Redirect to a zap channel ........
+
+2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31
+ Jan 2008) | 2 lines Add more missing locking of the agents list
+ ... ........
+
+ * /, channels/chan_agent.c: Merged revisions 101413-101414 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008)
+ | 2 lines Add missing locking to the find_agent() function.
+ ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31
+ Jan 2008) | 3 lines Move the locking from find_agent() into the
+ agent dialplan function handler to ensure that the agent doesn't
+ disappear while we're looking at it. ........
+
+2008-01-31 15:36 +0000 [r101393] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_realtime.c: Add missing braces. (closes issue #11886)
+ Reported by: sergee Patches: func_realtime_fix-r101392.diff
+ uploaded by sergee (license 138)
+
+2008-01-31 05:28 +0000 [r101373] Russell Bryant <russell@digium.com>
+
+ * CHANGES: remove entry that is no longer in the tree
+
+2008-01-30 23:10 +0000 [r101344] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: The deprecation of "username" in favor of
+ "defaultuser" for SIP peers unfortunately broke realtime
+ configurations which still used the "username" field. This was
+ taken care of properly when reading from realtime but was not
+ handled properly when updating a realtime peer. This change also
+ adds a deprecation NOTICE CLI message that will print if using
+ the deprecated "username" field. (closes issue #11880) Reported
+ by: cabal95 Patches: 11880.patch uploaded by putnopvut (license
+ 60) Tested by: cabal95
+
+2008-01-30 20:08 +0000 [r101322] Olle Johansson <oej@edvina.net>
+
+ * configs/cli.conf.sample: Clarify configuration file that can be
+ misunderstood
+
+2008-01-30 19:03 +0000 [r101296] Jason Parker <jparker@digium.com>
+
+ * apps/app_controlplayback.c: Allow disabling the default
+ ffwd/rewind keys in the ControlPlayback application. This is done
+ in a backward compat way. If the "default" key for ffwd/rew is
+ used for another option (such as stop), the "default" is removed.
+ (closes issue #11754) Reported by: johan Patches:
+ app_controlplayback.c.option3.patch uploaded by johan (license
+ 334) Tested by: johan, qwell
+
+2008-01-30 17:12 +0000 [r101271] Olle Johansson <oej@edvina.net>
+
+ * configs/rtppage.conf.sample (removed), apps/app_rtppage.c
+ (removed): Removing applications that wasn't ready for svn trunk,
+ as trunk now has pre-release status.
+
+2008-01-30 16:54 +0000 [r101269] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Make the VoicemailUsersList AMI command
+ consistent with other manager list functions. (closes issue
+ #11874) Reported by: srt Patches: voicemail_ami-11847.patch
+ uploaded by srt (license 378)
+
+2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change
+ function to void because it always returned the same value and no
+ one read it.
+
+ * main/rtp.c: Formatting fixes
+
+2008-01-30 15:42 +0000 [r101224] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_rtppage.c: Get trunk to compile
+
+2008-01-30 15:42 +0000 [r101223] Joshua Colp <jcolp@digium.com>
+
+ * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4
+ lines Fix an issue where if a frame of higher sample size
+ preceeded a frame of lower sample size and ast_slinfactory_read
+ was called with a sample size of the combined values or higher a
+ crash would happen. (closes issue #11878) Reported by: stuarth
+ ........
+
+2008-01-30 15:36 +0000 [r101221] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Update CHANGES with rtppage
+
+2008-01-30 15:35 +0000 [r101220] Jason Parker <jparker@digium.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 101219 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37
+ -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use
+ descending channel order of groups, rather than ascending. Fixes
+ a potential source of confusion in glare-type situations. Issue
+ 11875, reported by JimVanM. ........
+
+2008-01-30 15:30 +0000 [r101218] Olle Johansson <oej@edvina.net>
+
+ * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added):
+ Add rtppage() application to do multicast or unicast RTP paging
+ to SIP phones. (closes issue #11797) Reported by: macbrody
+ Patches: app_rtppage-20080130.c uploaded by macbrody (license
+ 352)
+
+2008-01-30 15:27 +0000 [r101217] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan
+ 2008) | 5 lines Fix a logic error with regards to autofill. Prior
+ to this change, it was possible for a caller to go out of turn if
+ autofill were enabled and callers ahead in the queue were
+ attempting to call a member. This change fixes this. ........
+
+2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: simplify this code and eliminate the return
+ value cast that is no longer necessary
+
+2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, include/asterisk/channel.h: Constifying the
+ interface to get pvt_ids in the bridge, based on suggestion from
+ (const char *) Kevin. Thanks!
+
+ * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7
+ lines Stop musiconhold on attended transfer. (closes issue
+ #11872) Reported by: gareth Patches: svn-101018.patch uploaded by
+ gareth (license 208) ........
+
+2008-01-30 00:58 +0000 [r101126] Jason Parker <jparker@digium.com>
+
+ * CHANGES: Fix a typo
+
+2008-01-30 00:04 +0000 [r101082] Russell Bryant <russell@digium.com>
+
+ * CHANGES, apps/app_speech_utils.c: Add the 'n' option to
+ SpeechBackground, which has the application not answer the
+ channel if it has not already been answered. (closes SPD-51)
+
+2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, build_tools/make_version: Merged revisions 101080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29
+ Jan 2008) | 1 line updated build_tools to handle the autotag
+ directory structure changes; changes related to BE-353. Patch by
+ The Russell and reviewed by The Me. ........
+
+2008-01-29 23:02 +0000 [r101036] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan
+ 2008) | 3 lines Remove a memory leak from updating realtime
+ queues ........
+
+2008-01-29 22:04 +0000 [r101018] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_curl.c: Oops, a sizeof error
+
+2008-01-29 19:41 +0000 [r100974] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan
+ 2008) | 6 lines Fixing an erroneous return value returned when
+ attempting to pause or unpause a queue member fails. Fixes
+ BE-366, thanks to John Bigelow for writing the patch. ........
+
+2008-01-29 17:58 +0000 [r100935] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 100934 via svnmerge ........ r100934 | file
+ | 2008-01-29 13:57:05 -0400 (Tue, 29 Jan 2008) | 4 lines Don't
+ forget to record the channel so we know whether it is bridged or
+ not later. (closes issue #11811) Reported by: slavon ........
+
+2008-01-29 17:44 +0000 [r100933] Russell Bryant <russell@digium.com>
+
+ * /, main/Makefile: Merged revisions 100932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008)
+ | 4 lines Fix the last couple of issues related to building from
+ a path that contains spaces. (closes issue #11834) ........
+
+2008-01-29 17:42 +0000 [r100931] Jason Parker <jparker@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan
+ 2008) | 6 lines Initialize an array to 0s if config option not
+ specified. (closes issue #11860) Patches:
+ misdn_get_config.v1.diff uploaded by IgorG (license 20) ........
+
+2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 100922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008)
+ | 3 lines Use GNU make magic instead of shell magic to escape
+ spaces in the working directory. (related to issue #11834)
+ ........
+
+ * Makefile, /: Merged revisions 100882 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008)
+ | 6 lines Fix building Asterisk when the working path has spaces
+ in it. (closes issue #11834) Reported by: spendergrass Patched
+ by: me ........
+
+2008-01-29 16:14 +0000 [r100843] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) |
+ 5 lines Allow zap groups above 30 to work properly. (closes issue
+ #11590) Reported by: tbsky ........
+
+2008-01-29 15:30 +0000 [r100833] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make externip work as documented. If no port
+ is specified it will use the value of bindport instead of always
+ being 5060. (closes issue #11858) Reported by: hmodes
+
+2008-01-29 10:50 +0000 [r100794-100795] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29
+ Jan 2008) | 1 line fixed potential segfault in misdn show
+ channels CLI command ........
+
+ * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03
+ Jan 2008) | 1 line make sure frame is completely clean, before we
+ send it to asterisk as DTMF. If we don't make it clean, it
+ happens that one way audio occurs.. ........
+
+2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8
+ lines (closes issue #11736) Reported by: MVF Patches:
+ bug11736-2.diff uploaded by oej (license 306) Tested by: oej,
+ MVF, revolution (russellb: This was the showstopper for the
+ release.) ........
+
+ * channels/chan_sip.c: Removing code that wasn't supposed to be
+ there at all, only at an experimental stage before I found
+ another solution. Thanks Kevin, for reminding me.
+
+2008-01-28 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-beta2 released.
+
+2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker@digium.com>
+
+ * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added),
+ doc/tex/channelvariables.tex, makeopts.in: Reintroduce more
+ chan_vpb stuff that was removed in r100421 and r100422
+
+2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/Makefile: Re-inserting chan_vpb into trunk.
+
+2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 100675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008)
+ | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went
+ to 't' instead of 'T') ........
+
+2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43
+ -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make
+ sure we put greeting files into the database like we do with the
+ others. Issue #11795 Reported by: dimas Patches: vmgreet.patch
+ uploaded by dimas (license 88) ........
+
+2008-01-28 20:58 +0000 [r100674] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 100673 via svnmerge ........ r100673 |
+ mmichelson | 2008-01-28 14:55:56 -0600 (Mon, 28 Jan 2008) | 3
+ lines Undoing the deprecation of chan_vpb. It is alive and well.
+ ........
+
+2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix up some T38 state change issues. (closes
+ issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch
+ uploaded by dimas (license 88)
+
+ * channels/chan_sip.c: Fix up two scheduling issues. In one
+ instance a scheduled item was not deleted when it should have
+ been and in the other it was scheduled again when it shouldn't
+ have been.
+
+2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge
+ of this commit was a NoOp, since res_features doesn't exist in
+ trunk. Thanks to qwell for pointing it out!
+
+ * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008)
+ | 5 lines For some reason, the use of this strdupa() is leading
+ to memory corruption on freebsd sparc64. This trivial workaround
+ fixes it. (closes issue #10300, closes issue #11857, reported by
+ mattias04 and Home-of-the-Brave) ........
+
+2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/logger.c: Normalize the detection for execinfo, so that
+ Linux (glibc) and other platforms with libexecinfo will generate
+ inline stack backtraces correctly.
+
+2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell@digium.com>
+
+ * /: Merged revisions 100626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008)
+ | 7 lines Fix a crash in ast_masq_park_call() (issue #11342)
+ Reported by: DEA Patches: res_features-park.txt uploaded by DEA
+ (license 3) ........
+
+2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) |
+ 1 line Correct a comment which made little/no sense. ........
+
+2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /,
+ include/asterisk/channel.h: Merged revisions 100581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28
+ Jan 2008) | 9 lines Make some deadlock related fixes. These bugs
+ were discovered and reported internally at Digium by Steve Pitts.
+ - Fix up chan_local to ensure that the channel lock is held
+ before the local pvt lock. - Don't hold the channel lock when
+ executing the timing function, as it can cause a deadlock when
+ using chan_local. This actually changes the code back to be how
+ it was before the change for issue #10765. But, I added some
+ other locking that I think will prevent the problem reported
+ there, as well. ........
+
+ * main/pbx.c: Clean up some formatting, and simplify a bit of code
+ using ast_str
+
+2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't do a network byte order conversion
+ when setting the socket's port variable to that of bindaddr's. It
+ is already in the correct network byte order. (closes issue
+ #11800) Reported by: hmodes
+
+2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert
+ some spaces to tabs
+
+ * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few
+ little formatting changes
+
+ * main/channel.c: These readlocks always fail for me on my mac, and
+ I saw it happen again today on another mac. We ignore the return
+ value of locking operations almost everywhere in Asterisk. So,
+ ignore these, as well, so Asterisk will actually work on systems
+ where this is occurring while I look into what the issue is.
+
+2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sched.h,
+ channels/chan_iax2.c: With the switch to the ast_sched_replace*
+ API in trunk, we lose the correction that was just merged from
+ 1.4, so this is a changeover to those APIs to use the macro
+ versions, so that we properly detect errors from ast_sched_del,
+ instead of simply ignoring the return values.
+
+ * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /,
+ channels/chan_sip.c, channels/chan_h323.c,
+ include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c,
+ channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged
+ revisions 100465 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008)
+ | 11 lines When deleting a task from the scheduler, ignoring the
+ return value could possibly cause memory to be accessed after it
+ is freed, which causes all sorts of random memory corruption.
+ Instead, if a deletion fails, wait a bit and try again (noting
+ that another thread could change our taskid value). (closes issue
+ #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan,
+ stuarth` ........
+
+2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker@digium.com>
+
+ * doc/tex/channelvariables.tex: Get rid of that last little bit.
+
+ * build_tools/menuselect-deps.in, configs/vpb.conf.sample
+ (removed), makeopts.in: Remove more remnants of chan_vpb
+
+2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc (removed), configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/Makefile, .cleancount: Removing chan_vpb from the tree
+
+ * /: Blocked revisions 100418 via svnmerge ........ r100418 |
+ mmichelson | 2008-01-25 16:32:41 -0600 (Fri, 25 Jan 2008) | 4
+ lines Deprecating chan_vpb. It is now preferred that users of
+ Voicetronix products use chan_zap in combination with their
+ zaptel drivers. ........
+
+2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) |
+ 2 lines This would have never been true, since we're passing
+ (sizeof(req.data) - 1) as the len to recvfrom(). ........
+
+2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c: correct a real problem and silence an annoying
+ compiler warning
+
+2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Insure that we are not going to pass a NULL
+ pointer to add_to_interfaces. (closes issue #11840) Reported by:
+ junky
+
+2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp@digium.com>
+
+ * main/dial.c, include/asterisk/dial.h: Add an API call that steals
+ the answered channel so that a destruction of the dialing
+ structure does not hang it up.
+
+2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as
+ the default, too
+
+2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/app.h: ummm... might be good if this macro
+ argument was actually used :-)
+
+ * include/asterisk/app.h: add the ability to define a structure
+ type for argument parsing when it would be useful to be able to
+ pass it between functions
+
+2008-01-24 22:02 +0000 [r100266] James Golovich <james@gnuinter.net>
+
+ * channels/chan_sip.c: Fix simple whitespace issue
+
+2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24
+ Jan 2008) | 2 lines make these macros not assume that the only
+ other field in the structure is 'argc'... this is true when
+ someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
+ to define your own structure as long as it has the right fields
+ ........
+
+2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp@digium.com>
+
+ * main/features.c: Minor cosmetic change...
+
+2008-01-24 18:35 +0000 [r100224] James Golovich <james@gnuinter.net>
+
+ * main/astmm.c: Increase the size of filenames stored when astmm is
+ used. If the path length was long they would be truncated and
+ grouped together with whatever matches
+
+2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp@digium.com>
+
+ * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp
+ branch. This adds a strictrtp option to rtp.conf which drops
+ packets that do not come from the remote party. (closes issue
+ #8952) Reported by: amorsen
+
+2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 100164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008)
+ | 2 lines Update main Asterisk copyright info to 2008 ........
+
+2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker@digium.com>
+
+ * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) |
+ 6 lines Fix compilation on Solaris. (closes issue #11832)
+ Patches: bug-11832.diff uploaded by snuffy (license 35) ........
+
+ * channels/chan_sip.c, main/features.c: Move chan_local dependency
+ into places (only one) that previously depended on res_features,
+ and used local channels
+
+2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
+ channels/chan_mgcp.c: Remove dependency on res_features from some
+ channel drivers. It is now part of the core and no longer exists
+ as a module.
+
+ * main/channel.c: Some more cosmetic changes.
+
+ * main/channel.c: Add some spacing.
+
+ * main/dial.c: Test hopefully over.
+
+ * main/dial.c: Testing something...
+
+2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix flag bit definitions to make code from
+ issue #11049 actually work; along the way, clarify comments and
+ add some dummy flag definitions for other multi-bit flags to
+ hopefully stop this from happening in the future (closes issue
+ #11049)
+
+2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c (removed), main/Makefile, main/features.c
+ (added), include/asterisk/_private.h, CHANGES, .cleancount,
+ main/asterisk.c, main/loader.c, include/asterisk/features.h: Move
+ code from res_features into (new file) main/features.c
+
+2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell@digium.com>
+
+ * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your
+ contributions!
+
+2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7
+ lines Second attempt. Don't change invitestate when receiving 18x
+ messages in CANCEL state. (issue #11736) Reported by: MVF Patch
+ by oej. ........
+
+ * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9
+ lines Make sure we don't cancel destruction on calls in CANCEL
+ state, even if we get 183 while waiting for answer on our CANCEL.
+ (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by
+ oej (license 306) Tested by: MVF ........
+
+2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23
+ Jan 2008) | 3 lines Fixing a typo. ........
+
+2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) |
+ 8 lines ChanSpy issues a beep when it starts at the beginning of
+ a list of channels to potentially spy on. However, if there were
+ no matching channels, it would beep at you over and over, which
+ is pretty annoying. Now, it will only beep once in the case that
+ there are no channels to spy on, but it will still beep again
+ once it reaches the beginning of the channel list again. (closes
+ issue #11738, patched by me) ........
+
+ * main/tcptls.c: Fix tcptls build when openssl isn't installed
+ (closes issue #11813) Reported by: tzafrir Patches:
+ asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: since echo canceler parameters in Zaptel are
+ now signed integers, allow them during parsing
+
+2008-01-23 16:21 +0000 [r99879] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 99878 via svnmerge ........ r99878 |
+ mmichelson | 2008-01-23 10:18:04 -0600 (Wed, 23 Jan 2008) | 4
+ lines These flag tests were illogical. They were testing sip_peer
+ flags on a sip_pvt. Thanks to Russell for helping to get this odd
+ problem figured out. ........
+
+2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Progress messages don't work (closes issue
+ #10497) Reported by: pj Patches: h323-announces-r99483.diff
+ uploaded by sergee (license 138) Tested by: pj
+
+2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure
+ that we are aware of a pending INVITE even if we're using TCP
+
+2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Coding guidelines fixups
+
+ * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008)
+ | 8 lines When we reset the password via an external command, we
+ should also reset the password stored in the in-memory list, too
+ (otherwise it doesn't really take effect). (closes issue #11809)
+ Reported by: davetroy Patches: fix_externpass.diff uploaded by
+ davetroy (license 384) ........
+
+ * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Oops, should have checked for a NULL obj, here, too
+ ........
+
+ * res/res_config_ldap.c: Coding guidelines cleanup
+
+ * /, main/acl.c: Merged revisions 99718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Just confirmed that all current platforms need this
+ header file ........
+
+ * /: Oops
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, doc/ldap.txt (added),
+ configure.ac, configs/res_ldap.conf.sample (added),
+ res/res_config_ldap.c (added), CHANGES, makeopts.in,
+ contrib/scripts/asterisk.ldap-schema (added),
+ contrib/scripts/asterisk.ldif (added): Add res_config_ldap for
+ realtime LDAP engine. (closes issue #5768) Reported by: mguesdon
+ Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon
+ (license 121) res_ldap.conf.sample uploaded by suretec (license
+ 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70)
+ asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested
+ by: oej, mguesdon, suretec, cthorner
+
+2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4
+ lines Thanks to Russell's education I realize that BUFSIZ has
+ changed since I learned the C language over 20 years ago...
+ Resetting chan_sip to the size of BUFSIZ that I expected in my
+ old head to avoid too heavy memory allocations on some systems.
+ ........
+
+ * doc/tex/channelvariables.tex, CHANGES: Documentation updates for
+ BRIDGEPVTCALLID
+
+2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 99643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Fix the defines for OS X (and Solaris, too) ........
+
+2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Make sure the command is not just present but is
+ also configured to be executed
+
+2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej@edvina.net>
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
+ Add a generic function to set the bridged call PVT unique id
+ string as a channel variable BRIDGEPVTCALLID This is important
+ for call tracing in log files and CDRs, so that the SIP callID
+ can be traced along servers. The CHANNEL dialplan function won't
+ work here, since the outbound channel is gone when we need the
+ Call-ID. Other channel drivers may now implement the same
+ function :-), but this patch only supports chan_sip.so. Inspired
+ by (issue #11816) Reported by: ctooley Patch by oej
+
+2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell@digium.com>
+
+ * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change
+ the Asterisk CLI startup commands feature to read commands to run
+ from cli.conf after a discussion on the -dev list.
+
+2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c, /, res/res_features.c,
+ channels/chan_agent.c, apps/app_followme.c: Merged revisions
+ 99594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3
+ lines Add more dependencies on chan_local and add a note to the
+ description of chan_local so that people don't disable it in
+ menuselect just to clean up. ........
+
+ * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5
+ lines Add dependency on chan_local to app_dial. Dial still runs
+ without chan_local, but will be missing forwarding functionality.
+ ........
+
+2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 99540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008)
+ | 7 lines Ensure that we can get an address even when we don't
+ have a default route. (closes issue #9225) Reported by: junky
+ Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
+ (license 14) Tested by: oej, loloski, sergee ........
+
+2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Point out a bug in some debug counter
+ handling
+
+2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add authentication options to the SIP
+ dialstring. Documentation follows separately (issue #11587)
+ Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by
+ sobomax (license 359)
+
+ * configs/sip.conf.sample: Documentation updates
+
+ * doc/siptls.txt: Small fixes
+
+ * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c,
+ include/asterisk/tcptls.h: Doxygen updates
+
+2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 99426 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21
+ Jan 2008) | 12 lines Fixing an issue wherein monitoring local
+ channels was not possible. During a channel masquerade, the
+ monitors on the two channels involved are swapped. In 99% of the
+ cases this results in the desired effect. However, if monitoring
+ a local channel, this caused the monitor which was on the local
+ channel to get moved onto a channel which is immediately hung up
+ after the masquerade has completed. By swapping the monitors
+ prior to the masquerade, we avoid the problem by tricking the
+ masquerade into placing the monitor back onto the channel where
+ we want it. During the investigation of the issue, the channel's
+ monitor was the only thing that was swapped in such a manner
+ which did not make sense to have done. All other variable
+ swapping made sense. ........
+
+2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Fix distinctive ring detection. Reported by:
+ milazzo Patches: drings.diff uploaded by milazzo (license 383)
+ Closes issue #11799
+
+2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Adding the
+ QUEUENAME variable to the variables set using the setqueuevar
+ option in queues.conf. Suggestion comes from Shaun2222 on IRC.
+
+2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_console.c: Remove compiler warning for
+ uninitialized variable
+
+ * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was
+ committed without any doxygen obviously. Tss tss.
+
+ * channels/chan_sip.c: Updating doxygen
+
+2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, /, res/res_odbc.c,
+ configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21
+ Jan 2008) | 8 lines Permit the user to specify number of seconds
+ that a connection may remain idle, which fixes a crash on
+ reconnect with the MyODBC driver. (closes issue #11798) Reported
+ by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak ........
+
+2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4
+ lines Bump the buffer size for Via headers up to 512. There are
+ some exceptionally large Via headers out there. (closes issue
+ #11783) Reported by: ofirroval ........
+
+2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej@edvina.net>
+
+ * CREDITS: Update
+
+2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Change over to using ast_debug so these
+ debug messages don't always show up.
+
+2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c: Add a "console active" CLI command,
+ which lets you find out which console device is currently active
+ for the Asterisk CLI, or to set it. Also, knock multiple device
+ support off of the to-do list.
+
+ * configs/console.conf.sample: correct the name of a CLI command
+ for getting available device names
+
+ * configs/console.conf.sample, channels/chan_console.c: Merge
+ changes from team/russell/console_devices - Add support for
+ multiple devices. All devices are configured in console.conf. -
+ Add "console list devices" CLI command to show configured
+ devices. Also, changed the old "list devices" to be "list
+ available", which queries PortAudio for all audio devices that
+ are available for use.
+
+ * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) |
+ 4 lines Fix a couple of memory leaks with frame handling.
+ Specifically, ast_frame_free() needed to be called on the frame
+ that came from the translator to signed linear. ........
+
+ * README: Add Cygwin as an "other" platform that is supported
+
+ * README: Various README updates
+
+2008-01-18 22:58 +0000 [r99128] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 99127 via svnmerge ........ r99127 | file |
+ 2008-01-18 18:57:15 -0400 (Fri, 18 Jan 2008) | 2 lines Remove the
+ __ in front of the unused variable. This causes some compilers to
+ freak out. ........
+
+2008-01-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-beta1 released.
+
+2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell@digium.com>
+
+ * CREDITS, include/asterisk/http.h, main/tcptls.c (added),
+ main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
+ main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
+ configs/sip.conf.sample, CHANGES: Merge changes from
+ team/group/sip-tcptls This set of changes introduces TCP and TLS
+ support for chan_sip. There are various new options in
+ configs/sip.conf.sample that are used to enable these features.
+ Also, there is a document, doc/siptls.txt that describes some
+ things in more detail. This code was implemented by Brett Bryant
+ and James Golovich. It was reviewed by Joshua Colp and myself. A
+ number of other people participated in the testing of this code,
+ but since it was done outside of the bug tracker, I do not have
+ their names. If you were one of them, thanks a lot for the help!
+ (closes issue #4903, but with completely different code that what
+ exists there.)
+
+ * main/frame.c, /, include/asterisk/translate.h: Merged revisions
+ 99081 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) |
+ 9 lines Revert adding the packed attribute, as it really doesn't
+ make sense why that would do any good. Fix the real bug, which is
+ to do the check to see if the frame came from a translator at the
+ beginning of ast_frame_free(), instead of at the end. This
+ ensures that it always gets checked, even if none of the parts of
+ the frame are malloc'd, and also ensures that we aren't looking
+ at free'd memory in the case that it is a malloc'd frame. (closes
+ issue #11792, reported by explidous, patched by me) ........
+
+ * /, include/asterisk/translate.h: Merged revisions 99079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) |
+ 4 lines Since we're relying on the offset between the frame and
+ the beginning of the translator pvt struct, set the packed
+ attribute to make sure we get to the right place. (potential fix
+ for issue #11792) ........
+
+2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: This should at least temporarily fix a
+ problem where the 't' Dial option is incorrectly passed to the
+ transferee when built-in attended transfers are used. There is
+ still a problem with 'T', but better to fix some problems than no
+ problems while we work on it. (closes issue #7904) Reported by:
+ k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee
+ (license 138) Tested by: sergee, otherwiseguy
+
+2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for
+ speed (closes issue #10723) Reported by: mnicholson Patches:
+ func-odbc-direct-execute1.diff uploaded by mnicholson (license
+ 96) Tested by: Corydon76, mnicholson, falves11
+
+ * res/res_odbc.c: Permit username and password to be NULL (which
+ enables pass-through from the layer above). Reported by: lurcher
+ Patch by: tilghman (Closes issue #11739)
+
+ * funcs/func_cut.c: Reset default CUT delimiter back to '-'
+
+2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c: Make the output of "console list
+ devices" a bit prettier.
+
+ * channels/chan_console.c: List which devices are inputs and
+ outputs in "console list devices"
+
+ * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for
+ ast_best_codec()
+
+ * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
+ Merged revisions 99004 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) |
+ 10 lines Have IAX2 optimize the codec translation path just like
+ chan_sip does it. If the caller's codec is in our codec list,
+ move it to the top to avoid transcoding. (closes issue #10500)
+ Reported by: stevedavies Patches: iax-prefer-current-codec.patch
+ uploaded by stevedavies (license 184)
+ iax-prefer-current-codec.1.4.patch uploaded by stevedavies
+ (license 184) Tested by: stevedavies, pj, sheldonh ........
+
+2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue
+ #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded
+ by DEA (license 3)
+
+2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker@digium.com>
+
+ * Makefile, build_tools/cflags.xml, channels/chan_zap.c,
+ main/dsp.c, configs/zapata.conf.sample: Add several busy
+ detection related defines to menuselect. Allow better busy detect
+ debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and
+ BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches:
+ busydetect_enhancement.patch uploaded by agx (license 298)
+ busydetect-r94975.diff uploaded by sergee (license 138)
+ Additional changes/cleanup by me.
+
+2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: state_interface could be NULL, so use the
+ never-NULL cur->state_interface for this check
+
+ * apps/app_queue.c: Get the device state of the state interface
+ instead of the interface when creating a new queue member. Thanks
+ to Atis Lezdins for bringing this up on the Asterisk-Dev mailing
+ list.
+
+2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker@digium.com>
+
+ * /, configs/zapata.conf.sample: Merged revisions 98991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600
+ (Thu, 17 Jan 2008) | 4 lines Add a clarification about the
+ immediate= option of zapata.conf Issue 11784, patch by klaus3000.
+ ........
+
+2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: major
+ reliability and performance improvement in VWMI monitoring for
+ FXO ports (code by markster, me and dbailey)
+
+ * res/res_config_curl.c: resolve (valid) compiler warning about
+ variable that could be used before being initialized
+
+2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c, doc/tex/phoneprov.tex,
+ configs/phoneprov.conf.sample: Update res_phoneprov to default to
+ setting the SERVER variable to the IP the HTTP request for the
+ config came in on and the SERVER_PORT to the bindport setting in
+ sip.conf. I've left in the ability to override these options,
+ because I can't always guess how someone might decide to do
+ something weird with what is available to them--although needing
+ to is pretty unlikely. Documentation was updated to reflect
+ preference for not setting serveraddr, serveriface, or
+ serverport. Tested on Linux and OS X.
+
+2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature
+ works, by allowing it to be a column NOT logged into the
+ database. This will allow more granularity of a decision
+ evaluated in the dialplan, then takes effect when posting the
+ CDR.
+
+2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell@digium.com>
+
+ * CHANGES, main/asterisk.c: Add support for an easy way to
+ automatically execute some Asterisk CLI commands immediately at
+ startup. Any commands in the startup_commands file in the
+ Asterisk config diretory will get executed. (closes issue #11781)
+ Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt
+ uploaded by jamesgolovich (license 176) -- With some changes by
+ me.
+
+2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build
+ against <package>_LIB, per recommendations from Russell.
+
+2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: Info about res_config_curl
+
+2008-01-16 22:36 +0000 [r98983] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 98982 via svnmerge ........ r98982 | russell
+ | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines Add an
+ unused pointer to the ast_channel struct. This makes the
+ ast_channel structure retain the same size as it had in previous
+ 1.4 releases. Also, all of the offsets for members in the
+ structure are still the same (except for the two pointers that
+ got replaced for the new spy/whisper architecture.) ........
+
+2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_curl.c (added), main/utils.c: New module
+ res_config_curl (closes issue #11747) Reported by: Corydon76
+ Patches: res_config_curl.c uploaded by Corydon76 (license 14)
+ 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jmls
+
+2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell@digium.com>
+
+ * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the
+ changes from issue #10665 from the team/group/sip_session_timers
+ branch. This set of changes introduces SIP session timers support
+ (RFC 4028). In short, this prevents stuck SIP sessions that were
+ not properly torn down due to network or endpoint failures during
+ an established SIP session. To quote some of the documentation
+ supplied with the patch: "The SIP Session-Timers is an extension
+ of the SIP protocol that allows end-points and proxies to refresh
+ a session periodically. The sessions are kept alive by sending a
+ RE-INVITE or UPDATE request at a negotiated interval. If a
+ session refresh fails then all the entities that support Session-
+ Timers clear their internal session state. In addition, UAs
+ generate a BYE request in order to clear the state in the proxies
+ and the remote UA (this is done for the benefit of SIP entities
+ in the path that do not support Session-Timers)." (closes issue
+ #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by
+ rjain (license 226) chan_sip.c.diff uploaded by rjain (license
+ 226) sip.conf.sample.diff uploaded by rjain (license 226)
+ proc_422_rsp_comment.diff uploaded by rjain (license 226)
+ chan_sip.c.cache.diff uploaded by rjain (license 226)
+ chan_sip.memalloc uploaded by rjain (license 226)
+ chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches
+ tracked in team/group/sip_session_timers, with some additional
+ fixes by russell and oej. Tested by: jtodd, rjain, loloski
+
+2008-01-16 20:36 +0000 [r98974-98975] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 98973 via svnmerge ........ r98973 | file |
+ 2008-01-16 16:34:30 -0400 (Wed, 16 Jan 2008) | 2 lines Bump up
+ cleancount due to previous commit that changed the channel
+ structure. ........
+
+ * /: Blocked revisions 98972 via svnmerge ........ r98972 | file |
+ 2008-01-16 16:33:47 -0400 (Wed, 16 Jan 2008) | 2 lines Replace
+ current spy architecture with backport of audiohooks. This should
+ take care of current known spy issues. ........
+
+2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Partially revert r93898, because it broke the way netsnmp was
+ being detected. rizzo, do you want to discuss so we can rethink
+ this, or do you have another way?
+
+ * CHANGES: Add note about new update.log to CHANGES, by request of
+ jmls and further prodding by jsmith.
+
+ * Makefile, /: Add logging for 'make update' command (also fixes
+ updates in some places). Issue #11766, initial patch by jmls.
+
+2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6
+ lines Add missing NULLs at end of two ast_load_realtimes. (closes
+ issue #11769) Reported by: tequ Patches: chaniax.patch uploaded
+ by dimas (license 88) ........
+
+2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 98964 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16
+ Jan 2008) | 10 lines Fix a deadlock in chan_local in
+ local_hangup. There was contention because the local_pvt was held
+ and it was attempting to lock a channel, which is the incorrect
+ locking order. (closes issue #11730) Reported by: UDI-Doug
+ Patches: 11730.patch uploaded by putnopvut (license 60) Tested
+ by: UDI-Doug ........
+
+2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Make users list static
+
+2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp@digium.com>
+
+ * main/dial.c, /: Merged revisions 98960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6
+ lines Introduce a lock into the dialing API that protects it when
+ destroying the structure. (closes issue #11687) Reported by:
+ callguy Patches: 11687.diff uploaded by file (license 11)
+ ........
+
+ * /, main/rtp.c: Merged revisions 98958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4
+ lines Add two more SDP names for ulaw and alaw. (closes issue
+ #11777) Reported by: tootai ........
+
+ * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6
+ lines Don't drop the old record route information when dealing
+ with packets related to a reinvite. (closes issue #11545)
+ Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by
+ kebl0155 (license 356) ........
+
+ * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This
+ seems to come from way back when and I can't think of a reason
+ for it being here, plus it could cause needless DNS lookups.
+ (closes issue #10983) Reported by: jtodd
+
+2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found
+ this problem with running the expr2 parser on OSX. Make the
+ #defines come out the same between the parser & lexer.
+
+2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp@digium.com>
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac, makeopts.in: Merged revisions 98951 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan
+ 2008) | 4 lines Add autoconf logic for speexdsp. Later versions
+ use a separate library for some things so we need to use it if
+ present in codec_speex. (closes issue #11693) Reported by: yzg
+ ........
+
+2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) |
+ 11 lines Change a buffer in check_auth() to be a thread local
+ dynamically allocated buffer, instead of a massive buffer on the
+ stack. This fixes a crash reported by Qwell due to running out of
+ stack space when building with LOW_MEMORY defined. On a very
+ related note, the usage of BUFSIZ in various places in chan_sip
+ is arbitrary and careless. BUFSIZ is a system specific define. On
+ my machine, it is 8192, but by definition (according to google)
+ could be as small as 256. So, this buffer in check_auth was 16
+ kB. We don't even support SIP messages larger than 4 kB! Further
+ usage of this define should be avoided, unless it is used in the
+ proper context. ........
+
+2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
+ Add the "filter" keyword
+
+2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell@digium.com>
+
+ * main/translate.c, include/asterisk/translate.h: Clean up
+ something I did for ABI compatability in 1.4
+
+ * main/frame.c, /, main/translate.c, main/abstract_jb.c,
+ channels/chan_iax2.c, codecs/codec_zap.c,
+ include/asterisk/frame.h, main/rtp.c,
+ include/asterisk/translate.h: Merged revisions 98943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15
+ Jan 2008) | 25 lines Commit a fix for some memory access errors
+ pointed out by the valgrind2.txt output on issue #11698. The
+ issue here is that it is possible for an instance of a translator
+ to get destroyed while the frame allocated as a part of the
+ translator is still being processed. Specifically, this is
+ possible anywhere between a call to ast_read() and
+ ast_frame_free(), which is _a lot_ of places in the code. The
+ reason this happens is that the channel might get masqueraded
+ during this time. During a masquerade, existing translation paths
+ get destroyed. So, this patch fixes the issue in an API and ABI
+ compatible way. (This one is for you, paravoid!) It changes an
+ int in ast_frame to be used as flag bits. The 1 bit is still used
+ to indicate that the frame contains timing information. Also, a
+ second flag has been added to indicate that the frame came from a
+ translator. When a frame with this flag gets released and has
+ this flag, a function is called in translate.c to let it know
+ that this frame is doing being processed. At this point, the flag
+ gets cleared. Also, if the translator was requested to be
+ destroyed while its internal frame still had this flag set, its
+ destruction has been deffered until it finds out that the frame
+ is no longer being processed. Admittedly, this feels like a hack.
+ But, it does fix the issue, and I was not able to think of a
+ better solution ... ........
+
+2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4
+ lines Based on the boundary found move over the correct amount.
+ (closes issue #11750) Reported by: tasker ........
+
+ * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4
+ lines Accept "; boundary=" not just ";boundary=" in the multipart
+ mixed content type. (closes issue #11750) Reported by: tasker
+ ........
+
+2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
+ backupdeleted option to app_voicemail (closes issue #10740)
+ Reported by: ruffle Patches: app_voicemail.diff uploaded by
+ ruffle (license 201) 10740-voicemail.diff uploaded by qwell
+ (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: blitzrage, mvanbaak, qwell
+
+2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directory.c: Big improvement for app_directory. This
+ patch breaks the do_directory function up so that it is more
+ easily parsed by the human brain. It also fixes some errors. I'll
+ quote dimas from the original bug description: "app_directory
+ contained some duplicate code even before addition of 'm' option.
+ Addition of that option doubled amount of that code. Worst of
+ all, there are minor differences between these code block and
+ bugs caused by these differences. 1. There is a memory leak. In
+ the 'menu' mode, result of the convert(pos) function is not freed
+ while it should be. 2. In the 'menu' mode check for
+ OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result,
+ application works in the mode opposite to what user expect
+ (checking last name when user wants the first nd vice versa). 3.
+ select_item function plays message for user using res = func1()
+ || func2() || func3()... construct. This construct loses the
+ actual value returned by ast_waitstream() for example so at the
+ end, res does not contain digit user dialed while listening to
+ the message. 4. (also in 1.4) application announces entries from
+ voicemail.conf/realtime separately from entries from users.conf.
+ I see no reason why doing so instead of building combined list.
+ 5. Alot of duplicated code as already mentioned." This was tested
+ by dimas and I (I tested under valgrind). A word of caution: any
+ bug fixes that happen in app_directory in 1.4 will almost
+ certainly not merge cleanly into trunk as a result of this, but
+ it is well worth it. Huge thanks to dimas for this wonderful
+ submission. (closes issue #11744) Reported by: dimas Patches:
+ dir3.patch uploaded by dimas (license 88) Tested by: putnopvut,
+ dimas
+
+ * /: Blocked revisions 98849 via svnmerge ........ r98849 |
+ mmichelson | 2008-01-14 14:59:26 -0600 (Mon, 14 Jan 2008) | 4
+ lines Adding in appropriate unlocks for the locks I added. Thanks
+ to joetester on IRC for pointing this out. ........
+
+2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c: Make sure the user's manager secret exists, even
+ if it is blank. (closes issue #11749) Reported by: srt
+
+2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson@digium.com>
+
+ * CHANGES: Add description of TOUPPER and TOLOWER dialplan
+ functions to CHANGES.
+
+2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Add proper call forwarding (all and busy)
+ support for chan_skinny. Note: NoAnswer support is currently not
+ implemented, as it would take a significant amount of work to
+ figure out how to do correctly. Closes issue #11310, patches,
+ testing, and support by DEA, mvanbaak, and myself.
+
+2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell@digium.com>
+
+ * /, main/translate.c: Merged revisions 98774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) |
+ 3 lines Revert a change that introduces an unacceptable
+ performance hit and is causing memory leaks ... (from rev 97973)
+ ........
+
+2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Fix for potential crash with vmexten
+
+2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Merged revisions 98737 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan
+ 2008) | 3 lines Fixing another compilation error. I'm a bit off
+ today :( ........
+
+ * /: Blocked revisions 98734 via svnmerge ........ r98734 |
+ mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3
+ lines Oops. Last commit had compilation error. ........
+
+ * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan
+ 2008) | 8 lines Adding explicit defaults for missing options to
+ init_queue. This is necessary because if a user either removes or
+ comments one of these options and reloads their queues, the
+ option will not reset to its default, instead maintaining the
+ value from prior to the reload. Thanks to John Bigelow for
+ pointing this error out to me. ........
+
+2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Print out a warning when spaces are used in the
+ variable name in Set and MSet. It is extremely hard to debug this
+ issue so this should make it easier. (closes issue #11759)
+ Reported by: caio1982 Patches: setvar_space_warning1.diff
+ uploaded by caio1982 (license 22)
+
+ * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update
+ documentation. (closes issue #11763) Reported by: IgorG Patches:
+ docupd.v1.diff uploaded by IgorG (license 20)
+
+2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell@digium.com>
+
+ * apps/app_jack.c: Add another small option for the JACK app and
+ JACK_HOOK function. The 'n' option tells JACK not to start jackd
+ automatically if it is not already running. Otherwise, the
+ default is that jackd will get started for you if it isn't
+ running already.
+
+ * CHANGES: - Break up the Misc. section a bit with a new section
+ for Misc. New Modules - Change spacing a bit in some places for
+ consistent indentation
+
+ * CHANGES, apps/app_jack.c (added): Bring in the code from
+ team/russell/jack/. Add a new module, app_jack, which provides
+ interfaces to JACK, the Jack Audio Connection Kit
+ (http://www.jackaudio.org/). Two interfaces are provided; there
+ is a JACK() application, and a JACK_HOOK() function. Both
+ interfaces create an input and output JACK port. The application
+ makes these ports the endpoint of the call. The audio coming from
+ the channel goes out the output port and whatever comes back in
+ on the input port is what gets sent to the channel. The
+ JACK_HOOK() function turns on a JACK audiohook on the channel.
+ This lets you run the audio coming from a channel through JACK,
+ and whatever comes back in is what gets forwarded on as the
+ channel's audio. This is very useful for building custom vocoders
+ or doing recording or analysis of the channel's audio in another
+ application. In case anyone is curious, the platform that
+ inspired me to write this is PureData (http://puredata.info/). I
+ wrote these JACK interfaces so that I could use Pd to do
+ interesting things with the audio of phone calls ...
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+ configure script check for JACK.
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ Remove KDE configure script check that isn't used
+
+ * main/audiohook.c: Remove a duplicate lock of the audiohook lock
+ when destroying manipulate audiohooks. This causes an error when
+ we attempt to destroy the lock later when freeing the audiohook.
+
+ * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar",
+ which allows you to set a channel variable (or function) on an
+ active channel from the CLI.
+
+2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Conversion to load manager.conf into memory did
+ not convert the password functions correctly. (Closes issue
+ #11749)
+
+2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * /, main/http.c: merging a comment added in 1.4
+
+2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and
+ ensure that the AMI DNDState event always gets generated. (closes
+ issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded
+ by tzafrir (modified by me) (license 46)
+
+2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008)
+ | 4 lines Add a connection timeout attribute, as that was what
+ was intended with the login timeout, but ODBC divides it up into
+ 2 different timeouts. (Closes issue #11745) ........
+
+2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell@digium.com>
+
+ * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some
+ extra checking to help out with a potential error when trying to
+ run "make asterisk.pdf" when not all of the right packages are
+ installed. (closes issue #10763) Reported by: Corydon76 Patches:
+ 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
+ 'auto' signalling mode for Zaptel channels. (closes issue #11690)
+ Reported by: tzafrir Patches: signaling_to_signalling.diff
+ uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded
+ by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir
+ (license 46) zap_no_default_sig.diff uploaded by tzafrir (license
+ 46) zap_signal_auto.diff uploaded by tzafrir (license 46)
+
+2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp@digium.com>
+
+ * main/event.c: Goodbye again drumkilla.
+
+ * main/event.c: drumkilla ftw.
+
+ * main/audiohook.c: I am no longer Rockin'
+
+ * main/audiohook.c: Testing something...
+
+2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) |
+ 9 lines Fix up setting the EID on BSD based systems. (closes
+ issue #11646) Reported by: caio1982 Patches:
+ dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
+ dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
+ by: caio1982, mvanbaak ........
+
+2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 98325 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6
+ lines If the incoming RTP stream changes codec force the bridge
+ to break if the other side does not support it. (closes issue
+ #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch
+ uploaded by tsearle (license 373) ........
+
+ * /, res/res_agi.c: Merged revisions 98317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6
+ lines If the channel is hungup during RECORD FILE send a result
+ code of -1 to be uniform with everything else. (closes issue
+ #11743) Reported by: davevg Patches: res_agi.diff uploaded by
+ davevg (license 209) ........
+
+2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 98315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan
+ 2008) | 5 lines Properly report the hangup cause as no answer
+ when someone does not answer (closes issue #10574, reported by
+ boch, patched by moy) ........
+
+2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell@digium.com>
+
+ * codecs/codec_resample.c: Kevin noted that the thing that I
+ _actually_ changed here was that I converted a value from a
+ double, to a float, back to a double. Sure enough, when I changed
+ my interim variable back to a double, it still blows up.
+ Switching all of these to a float fixes the problem. This seems
+ like a compiler bug where a double passed as an argument isn't
+ getting properly aligned, so I'll have to see if I can replicate
+ it with a small test program. (related to issue #11725)
+
+ * codecs/codec_resample.c: Fix a bus error that happened when
+ asterisk was built with optimizations on with platforms that
+ explode on unaligned access. I'm not exactly sure why this fixes
+ it, but it fixed it on the machine I was testing on. If it makes
+ sense to you, feel free to enlighten me. :) (closes issue #11725,
+ patched by me)
+
+2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to
+ adaptive_odbc, too
+
+ * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix
+
+2008-01-11 18:26 +0000 [r98267] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 98265 via svnmerge ........ r98265 | russell
+ | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines
+ Backport the ability to set the ToS bits on Linux when not
+ running as root. Normally, we would not backport features into
+ 1.4, but, I was convinced by the justification supplied by the
+ supplier of this patch. He pointed out that this patch removes a
+ requirement for running as root, thus reducing the potential
+ impacts of security issues. (closes issue #11742) Reported by:
+ paravoid Patches: libcap.diff uploaded by paravoid (license 200)
+ ........
+
+2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4
+ lines Ensure the return value of ast_bridge_call is passed back
+ up as the application return value. This is needed for transfers
+ to function so the PBX core knows to continue execution. (closes
+ issue #10327) Reported by: kkiely ........
+
+2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell@digium.com>
+
+ * codecs/codec_g722.c: At one point during working on this module,
+ I had the lin/lin16 versions of the framein callbacks different.
+ However, they are now the same again, so remove the duplicate
+ code and use the same functions for the lin/lin16 versions.
+
+2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008)
+ | 2 lines Back out changes from revision 97077, since it wasn't
+ perfect ........
+
+ * doc/manager_1_1.txt: Documentation updates
+
+2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO
+ using '!' as the 'digit'
+
+2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 98082 via svnmerge ........ r98082 | russell
+ | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines Fix
+ samples vs. length calculations for g722 ........
+
+ * codecs/codec_g722.c, main/frame.c: - Fix the last set of places
+ where incorrect assumptions were made about the sample length
+ with g722. It is _2_ samples per byte, not 1. This was all over
+ the place, and I believed it, and it is what caused me to take so
+ long to figure out what was broken. - Update copyright
+ information on codec_g722.
+
+2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson@digium.com>
+
+ * main/translate.c: Fix "core show translation" to not output
+ information for "unknown" codecs. This fix was made in favor of
+ the proposed patch since it doesn't involve changing a core codec
+ define. (closes issue #11722, reported and initially patched by
+ caio1982, final patch by me)
+
+2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new
+ global and per-peer option to chan_sip, qualifyfreq, which allows
+ you to set the qualify frequency. (closes issue #11597) Reported
+ by: wilder Patches: qualifyfreq5.patch uploaded by wilder
+ (license 362) -- with some mods by me
+
+ * /: Blocked revisions 98025 via svnmerge ........ r98025 | russell
+ | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines Simplify
+ this code with a suggestion from Luigi on the asterisk-dev list.
+ Instead of using is16kHz(), implement a format_rate() function.
+ ........
+
+ * main/translate.c: Simplify this code with a suggestion from Luigi
+ on the asterisk-dev list. Instead of using is16kHz(), implement a
+ format_rate() function.
+
+2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008)
+ | 6 lines 1) When we get a translated frame out, clone it,
+ because if the translator pvt is freed before we use the frame,
+ bad things happen. 2) Getting a failure from ast_sched_delete
+ means that the schedule ID is currently running. Don't just
+ ignore it. (Closes issue #11698) ........
+
+2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant <russell@digium.com>
+
+ * /, main/translate.c: Merged revisions 97976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) |
+ 3 lines Fix various timing calculations that made assumptions
+ that the audio being processed was at a sample rate of 8 kHz.
+ ........
+
+ * codecs/codec_g722.c: Fix various issues in codec_g722. - The most
+ common fix being made here is to fix all of the places where the
+ number of output samples and output bytes gets updated in the
+ translator state structure. - Fix a number of other places where
+ the number of samples provided as an initialization value to a
+ struct was incorrect.
+
+ * codecs/codec_resample.c: Fix the buffer_samples value. For signed
+ linear, the number of samples needed to fill the buffer is half
+ the buffer size.
+
+2008-01-10 21:58 +0000 [r97933] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan
+ 2008) | 6 lines Let us leave a voicemail for ourself if we have
+ logged into VoiceMailMain and chosen to leave a message. (closes
+ issue #11735, reported and patched by jamessan) ........
+
+2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged
+ revisions 97889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1
+ line Applied the same fixes for ael.flex as was done in 97849 for
+ ast_expr2.fl; overrode the normally generate yyfree func with our
+ own version that checks the pointer for non-null before passing
+ to free(). Also takes care of a little problem with 2.5.33 and
+ the use of the __STDC_VERSION__ macro. ........
+
+ * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged
+ revisions 97849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1
+ line This is a fix for 2 things: a problem Terry was having in
+ OSX with null pointers, which was my fault, as I probably forgot
+ to run the sed script last time I made mods. So, I moved the fix
+ into the flex input itself. Then, I found when I used flex
+ 2.5.33, that it was using __STDC_VERSION__, and that's not real
+ good; so I added back in a DIFFERENT sed script to fix that
+ little mess. Tested everything, a couple different ways. Hope I
+ did no harm, at the least. ........
+
+2008-01-10 20:13 +0000 [r97848] Jason Parker <jparker@digium.com>
+
+ * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan
+ 2008) | 1 line Fix a comment that is no longer true. ........
+
+2008-01-10 20:05 +0000 [r97846] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Use the appropriate line ending for the
+ X-Asterisk-VM-Message-Type header. (closes issue #11734, reported
+ and patched by jaroth)
+
+2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson <twilson@digium.com>
+
+ * main/ast_expr2f.c: heh, remove patch to generated file.
+
+ * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was
+ getting WARNINGS under MALLOC_DEBUG)
+
+2008-01-10 17:38 +0000 [r97805] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_odbc.c: Fix problem with timestr going out of scope
+ (Closes issue #11726, closes issue #11731)
+
+2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant <russell@digium.com>
+
+ * formats/format_sln16.c: minor formatting changes
+
+ * main/translate.c: spaces to tabs
+
+ * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK
+ check again. I changed this to an inline implementation to fix a
+ small bug, but after a discussion with rizzo, I went to change it
+ back. Also, it turns out that the implementation of the macro
+ already supported what was needed to fix the problem.
+
+ * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample,
+ pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) |
+ 2 lines Remove other remnants of pbx_kdeconsole ........
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+ pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) |
+ 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in
+ ages, and nobody has complained. (closes issue #11706, reported
+ by caio1982) ........
+
+2008-01-10 15:12 +0000 [r97698] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan
+ 2008) | 6 lines Don't try to copy the category from the group if
+ no category exists. (closes issue #11724) Reported by: IgorG
+ Patches: group_count.v1.patch uploaded by IgorG (license 20)
+ ........
+
+2008-01-10 00:54 +0000 [r97657] Russell Bryant <russell@digium.com>
+
+ * include/asterisk.h: These prototypes are not supposed to be in
+ asterisk.h. They are already in version.h.
+
+2008-01-10 00:50 +0000 [r97656] Steve Murphy <murf@digium.com>
+
+ * include/asterisk.h, channels/console_video.c, utils/astman.c,
+ channels/console_board.c, channels/vgrabbers.c: The fixes in this
+ commit are mainly to allow compiling of trunk with
+ --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly,
+ the version.c needs to be in the OBJS line; asterisk.h was chosen
+ to have the prototypes for ast_get_version, ast_get_version_num;
+ and the ASTERISK_FILE_VERSION macro needs to be used after
+ including asterisk.h in a few files. I hope I did the right
+ thing. If not, let me know.
+
+2008-01-10 00:39 +0000 [r97655] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: oops, missed the case of a 0 permission (which
+ should mean everybody is allowed, not nobody)
+
+2008-01-10 00:22 +0000 [r97653] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Attempt at making lookup_iface work under
+ FreeBSD. Not yet tested, but it compiles under OS X. And still
+ works under linux.
+
+2008-01-10 00:17 +0000 [r97652] Russell Bryant <russell@digium.com>
+
+ * codecs/Makefile: Fix this so it doesn't force codec_g722 to get
+ relinked every time
+
+2008-01-10 00:12 +0000 [r97651] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, main/manager.c, channels/chan_sip.c,
+ res/res_features.c, pbx/pbx_realtime.c,
+ configs/manager.conf.sample, CHANGES, channels/chan_iax2.c,
+ include/asterisk/manager.h, apps/app_stack.c, main/db.c,
+ apps/app_voicemail.c: Several manager changes: 1) Add the
+ Dialplan class, for NewExten and VarSet events, which should cut
+ down on the volume of traffic in the Call class. 2) Permit some
+ commands to be run from multiple classes, such as allowing DBGet
+ to be run from either the System or the Reporting class. 3)
+ Heavily document each class in the sample config, as there were
+ several that made no sense to be in the write= line, and two that
+ made no sense to be in the read= line (since they controlled no
+ permissions there). (Closes issue #10386)
+
+2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant <russell@digium.com>
+
+ * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the
+ files changes
+
+ * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) |
+ 2 lines Strip terminal sequences from the verbose messages
+ ........
+
+ * configure: re-gen configure
+
+ * configure.ac: re-add check for gtk1, which is used for
+ pbx_gtkconsole (related to issue #11706)
+
+ * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) |
+ 3 lines Make pbx_gtkconsole build ... but doesn't actually load
+ on my system still (related to issue #11706) ........
+
+2008-01-09 21:37 +0000 [r97634] Terry Wilson <twilson@digium.com>
+
+ * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml,
+ phoneprov/polycom.xml, res/res_phoneprov.c (added),
+ funcs/func_strings.c, phoneprov/000000000000-phone.cfg,
+ configs/modules.conf.sample, main/acl.c,
+ include/asterisk/localtime.h, CHANGES,
+ configs/phoneprov.conf.sample (added), Makefile, phoneprov
+ (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c,
+ doc/tex/asterisk.tex: Added a new module, res_phoneprov, which
+ allows auto-provisioning of phones based on configuration
+ templates that use Asterisk dialplan function and variable
+ substitution. It should be possible to create phone profiles and
+ templates that work for the majority of phones provisioned over
+ http. It is currently only intended to provision a single user
+ account per phone. An example profile and set of templates for
+ Polycom phones is provided. NOTE: Polycom firmware is not
+ included, but should be placed in AST_DATA_DIR/phoneprov/configs
+ to match up with the included templates.
+
+2008-01-09 20:30 +0000 [r97620-97623] Jason Parker <jparker@digium.com>
+
+ * /, main/cli.c: Merged revisions 97622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600
+ (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a
+ command could not be found. Also fix a comment which may have led
+ to this happening. Issue 11718, reported by kshumard. ........
+
+ * /, main/cli.c: Merged revisions 97618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1
+ line Fix some locking and return value funkiness. We really
+ shouldn't be unlocking this lock inside of a function, unless we
+ locked it there too. ........
+
+2008-01-09 18:53 +0000 [r97577] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan
+ 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some
+ smaller functions this time ........
+
+2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c: remove a wrong 'const'
+
+ * images/kpad2.jpg: add annotations for the two message windows we
+ use.
+
+2008-01-09 18:04 +0000 [r97531] Russell Bryant <russell@digium.com>
+
+ * /, res/res_features.c: Merged revisions 97529 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) |
+ 2 lines Fix saying the parking space number to the caller doing
+ the parking ... ........
+
+2008-01-09 18:03 +0000 [r97530] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/console_board.c,
+ channels/console_video.h: Two changes: - support scrolling of
+ message window; - simplify the code for creating a message
+ window, and try it using a second one in the top of the keypad
+ (where we echo the dialed number). The 'skin' that supports these
+ two windows will be committed separately.
+
+2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008)
+ | 2 lines report the same message whether Zaptel does not have
+ transcoder support loaded or no transcoders were found ........
+
+2008-01-09 16:59 +0000 [r97490] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09
+ Jan 2008) | 7 lines Set the caller id within the gtalk_alloc
+ function. As underlined in issue #10437 by Josh, we need to
+ prevent a possible memory leak. We only set the name part of the
+ caller id, the number part is not relevant when dealing with
+ JIDs. Closes issue #11549. ........
+
+2008-01-09 16:44 +0000 [r97488] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/console_video.c,
+ channels/console_board.c, channels/console_video.h: Implement
+ keyboard handling, and use it to enter a number to dial in the
+ 'message' area under the keypad. Now you can make calls using the
+ keypad as a regular phone (or the keyboard for chars not present
+ on the keypad)
+
+2008-01-09 16:13 +0000 [r97451] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6
+ lines Don't do conferencing totally in Zaptel if Monitor is
+ running on the channel. (closes issue #11709) Reported by:
+ BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license
+ 371) ........
+
+2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008)
+ | 2 lines pass the right variable to get an error string... oops
+ ........
+
+ * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008)
+ | 2 lines add error number output to ioctl failure messages to
+ help with debugging ........
+
+2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_video.c, channels/console_video.h: implement the
+ "console startgui" and "console stopgui" commands so you can
+ start and stop the gui even outside of a call. This is convenient
+ for testing, and also for using the keypad to pick up a call, and
+ to dial a number (the latter not yet implemented, but should be
+ close).
+
+ * channels/chan_oss.c: make get_video_desc() return the active
+ console if passed a null argument (channel).
+
+2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: New option in trunk, needs strdupa to be safe,
+ too
+
+ * /, main/editline/readline.c, main/cli.c: Merged revisions 97350
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008)
+ | 5 lines Allow filename completion on zero-length modules,
+ remove a memory leak, remove a file descriptor leak, and make
+ filename completion thread-safe. Patched and tested by tilghman.
+ (Closes issue #11681) ........
+
+2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan
+ 2008) | 3 lines use the \retval doxygen command properly ........
+
+ * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan
+ 2008) | 5 lines Part 1 of N of adding doxygen comments to
+ app_queue. I picked some of the most common functions used (which
+ also happen to be some the biggest/ugliest functions too) to
+ document first. I'm pretty new to doxygen so criticism is
+ welcome. ........
+
+2008-01-08 23:51 +0000 [r97305] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Add a new flag 'd' (with optional context)
+ permitting any extension within that context to be entered as a
+ new extension during the playback of a voicemail greeting. Patch
+ inspired by bluecrow76, by tilghman. (Closes issue #7063)
+
+2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_board.c: add copyright (most of this code was
+ written by Marta Carbone), remove some unused code, add/clarify
+ some comments.
+
+ * images/kpad2.jpg: Add the annotation for the textarea used for
+ messages, and also change the background from white to something
+ different to show that we can make use of fonts with transparent
+ background.
+
+ * images/font.png (added): add a font suitable for use with the
+ console GUI. The background of this particular image is
+ transparent so we can preserve the original background when we
+ draw strings.
+
+ * channels/console_gui.c, channels/console_video.c,
+ channels/console_board.c (added), channels/Makefile: add support
+ for textareas, used for various dialog windows on the gui. The
+ main code to implement the textarea is in console_board.c, and
+ uses a simple png image with the font, blitting characters on the
+ designated areas of the main screen. Additionally we provide some
+ annotations in the image used as a skin to indicate which areas
+ are used for text messages. (images will be committed
+ separately). At the moment the dialog area is only used to
+ display a running counter, just as a proof of concept.
+
+2008-01-08 21:56 +0000 [r97248] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c: Initialize new variable to NULL
+
+2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 97206 via svnmerge ........ r97206 |
+ mmichelson | 2008-01-08 15:24:48 -0600 (Tue, 08 Jan 2008) | 3
+ lines Some coding guidelines-related cleanup ........
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
+ option of specifying a second interface in a member definition
+ for a queue. app_queue will monitor this second device's state
+ for the member, even though it actually calls the first
+ interface. This ability has been added for statically defined
+ queue members, realtime queue members, and dynamic queue members
+ added through the CLI, dialplan, or manager. (closes issue
+ #11603, reported by acidv)
+
+2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_console.c: Change reference to external library so
+ it appears on the extref listing
+ http://www.asterisk.org/doxygen/trunk/extref.html
+
+ * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is
+ out with bug fixes.
+
+2008-01-08 20:56 +0000 [r97198] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008)
+ | 3 lines Increase constants to where we're less likely to hit
+ them while debugging. (Closes issue #11694) ........
+
+2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: One line documentation ftw!
+
+ * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6
+ lines Fix various DTMF issues in chan_mgcp. (closes issue #11443)
+ Reported by: eferro Patches:
+ dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
+ 337) ........
+
+2008-01-08 20:45 +0000 [r97193] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan
+ 2008) | 9 lines Making some changes designed to not allow for a
+ corrupted mailstream for a vm_state. 1. Add locking to the
+ vm_state retrieval functions so that no linked list corruption
+ occurs. 2. Make sure to always grab the persistent vm_state when
+ mailstream access is necessary. 3. Correct an incorrect return
+ value in the init_mailstream function. (closes issue #11304,
+ reported by dwhite) ........
+
+2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Move common code for setting T38
+ capabilities and fix a bug with fax detection in the SIP RTP read
+ callback. It's still sort of silly... but more on that later.
+ (closes issue #11239) Reported by: dimas Patches:
+ sipt38prop.patch uploaded by dimas (license 88)
+
+ * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan
+ 2008) | 4 lines If no group has been provided to the GROUP_COUNT
+ dialplan function then use the first one specific to the channel.
+ (closes issue #11077) Reported by: m4him ........
+
+2008-01-08 19:06 +0000 [r97125] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008)
+ | 3 lines Apply multiple crash fixes, found in issue #11386, but
+ not completely closing that issue. ........
+
+2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4
+ lines Make app_queue calls work with directed pickup. (closes
+ issue #11700) Reported by: jbauer ........
+
+ * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically
+ appear in extconf. (closes issue #11703) Reported by: dmartin
+
+2008-01-07 23:03 +0000 [r96988] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c: add support for cropping the keypad image
+ while displaying it. This way it can contain additional elements
+ (e.g. fonts, buttons, widgets) without having to use a zillion
+ files to store them.
+
+2008-01-07 22:31 +0000 [r96987] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Explicitly make literal constants long
+ where they are expected to be.
+
+2008-01-07 21:12 +0000 [r96936] Jason Parker <jparker@digium.com>
+
+ * main/config.c: Display a message if no config mappings are found
+ with "core show config mappings". Closes issue #11704, patch by
+ kshumard.
+
+2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Document some weird casting magic that's
+ necessary to interface with the c-client
+
+ * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding
+ user-configurable TCP timeout settings to IMAP voicemail. This
+ could go a long way towards preventing unexplainable hangs
+ experienced by people. In the case of MWI hangs, this also will
+ mean that the SIP port isn't blocked anymore. (closes issue
+ #11665, reported by yehavi)
+
+2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant <russell@digium.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 96932 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r96932 | russell | 2008-01-07 14:47:52 -0600
+ (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07
+ Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com
+ ........ ................
+
+ * configs/http.conf.sample: Add a note about viewing the default
+ set of documentation using the built-in http server
+
+ * Makefile: If the HTML documentation exists, install it in the
+ static-http/docs directory so that it can be viewed through the
+ Asterisk http server if it is turned on.
+
+ * build_tools/prep_tarball: Build the HTML version of the doc files
+ for tarballs, as well
+
+ * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) |
+ 3 lines Don't crash if something happens when setting up an SMDI
+ interface and it gets destroyed before the SMDI port handling
+ thread gets created. ........
+
+2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_sln16.c (added): add a file-format driver for
+ 16KHz signed linear... which may or may not work
+
+2008-01-07 15:52 +0000 [r96858] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck
+ manager commands from loader.c to manager.c. Previously they
+ would get registered twice because of the way manager.c operates.
+ (closes issue #11699) Reported by: caio1982 Patches:
+ manager_module_commands1.diff uploaded by caio1982 (license 22)
+
+2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c: update comments to reflect reality (or at
+ least planned behaviour). minor code cleanups
+
+ * channels/console_gui.c: resolve a load-time problem avoiding a
+ call to console_do_answer. On passing, fix dialling from the
+ keypad.
+
+2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant <russell@digium.com>
+
+ * res/snmp/agent.c: Convert this file over the new method of
+ getting the Asterisk version. (I don't have this building on this
+ machine, so caio1982 on IRC is going to test it for me. :) )
+
+ * Makefile, funcs/func_version.c, main/manager.c,
+ channels/chan_sip.c, main/Makefile, build_tools/make_version_c
+ (added), include/asterisk/version.h (added), res/res_agi.c, main,
+ main/http.c, build_tools/make_version_h (removed),
+ include/asterisk, main/asterisk.c: Now that the version.h file
+ was getting properly regenerated every time the svn revision
+ changed, every module that used the version was getting rebuilt
+ after every svn update. This severly annoyed me pretty quickly,
+ so I have improved the situation. Now, instead of generating
+ version.h, main/version.c is generated. version.c includes the
+ version information, as well as a couple of API calls for modules
+ to retrieve the version. So now, only version.c will get rebuilt,
+ and the main asterisk binary relinked, which is must faster than
+ rebuilding http.c, manager.c, asterisk.c, relinking the asterisk
+ binary, chan_sip.c, func_version.c, res_agi ... The only minor
+ change in behavior here is that the version information reported
+ by chan_sip, for example, is the version of the Asterisk core,
+ and not necessarily the Asterisk version that the chan_sip module
+ came from.
+
+ * main/pbx.c: Print out the name of a function being registered in
+ color, just like the name of applications when they get
+ registered.
+
+ * UPGRADE.txt: Add a note about changing modules.conf since another
+ console channel driver is now present that can not be used at the
+ same time as chan_alsa or chan_oss.
+
+ * channels/chan_console.c: Add the URL to the home page for
+ portaudio. Also add the location of the svn repository to check
+ out portaudio v19.
+
+ * /, main/devicestate.c: Merged revisions 96644 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) |
+ 2 lines Don't pass an empty string as the device name. ........
+
+2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_usbradio.c: improve chan_usbradio to use
+ indications just like chan_alsa/chan_oss do now
+
+2008-01-04 23:12 +0000 [r96576] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/devicestate.c: Merged revisions 96575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008)
+ | 7 lines Fix the problem of notification of a device state
+ change to a device with a '-' in the name. Could probably do with
+ a better fix in trunk, but this bug has been open way too long
+ without a better solution. Reported by: stevedavies Patch by:
+ tilghman (Closes issue #9668) ........
+
+2008-01-04 22:57 +0000 [r96574] Jason Parker <jparker@digium.com>
+
+ * /, res/res_features.c: Merged revisions 96573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600
+ (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if
+ using PARKINGEXTEN and the slot is full. Issue 11237, patch by
+ me. ........
+
+2008-01-04 19:35 +0000 [r96547] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008)
+ | 4 lines If you change the bindaddr in sip.conf to a non-bound
+ address and reload, sip goes kablooie. Reported and patched by:
+ one47 (Closes issue #11535) ........
+
+2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, acinclude.m4: [commit message] (closes issue
+ #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff
+ uploaded by tzafrir (license 46) (modified by me and added
+ configure script support)
+
+2008-01-04 17:19 +0000 [r96499] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over
+ unsecured network connections only. This authentication mechanism
+ is implemented under the iksemel API, which makes use of GnuTLS,
+ whereas we use OpenSSL. Note : there's ongoing dicsussion at the
+ SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
+ http://ietfreport.isoc.org/ids-wg-sasl.html.
+
+2008-01-04 16:21 +0000 [r96450] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) |
+ 7 lines Make use of the temporary channel pointer while the pvt
+ is unlocked. (closes issue #11675) Reported by: flefoll Patches:
+ chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll
+ (license 244) ........
+
+2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: we have to *always* use a completely silent 'make'
+ invocation for generating the module embedding rules
+
+ * Makefile: there was no reason to add this define for non-Solaris
+ platforms
+
+2008-01-03 22:46 +0000 [r96395] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) |
+ 3 lines Don't crash if the iax2 pvt structure has been destroyed
+ before we get to this point (closes issue #11672, reported by
+ snuffy, patched by me) ........
+
+2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/channel.h: Document recent API addition
+
+ * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03
+ Jan 2008) | 4 lines Missed initialization caused crash. Reported
+ and fixed by: tiziano (Closes issue #11671) ........
+
+ * main/channel.c: Allow the uniqueid to be used for searching for a
+ channel in the list. Reported and initially patched by:
+ michael-fig (Closes issue #11340)
+
+2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, tests/Makefile (added), tests/test_skel.c (added),
+ tests (added): add some simple infrastructure for modules to be
+ used for testing parts of Asterisk
+
+ * channels/answer.h (removed), channels/ring10.h (removed),
+ channels/busy.h (removed), channels/ringtone.h (removed),
+ channels/Makefile, channels/chan_oss.c, channels/gentone.c
+ (removed), channels: eliminiate sound_thread() and other stuff
+ from chan_oss since Asterisk indications can handle it remove
+ gentone and all the headers containing tones that are no longer
+ needed
+
+ * channels/chan_alsa.c: coding guidelines cleanup remove background
+ thread and all sound generation mechanisms, as the built-in
+ indications can handle everything that is needed
+
+2008-01-03 14:47 +0000 [r96221] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03
+ Jan 2008) | 1 line when overlapdial was used and no number was
+ dialed, the call was dropped, now we just jump into the s
+ extension, which makes a lot more sense. ........
+
+2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Add coordination between AMI and AGI applications,
+ with an asyncagi method Feature proposed and patched by: moy
+ (Closes issue #11282)
+
+ * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility
+ fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669)
+
+2008-01-02 23:48 +0000 [r96103] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan
+ 2008) | 4 lines We need to reset the membername to NULL on each
+ iteration of this loop, otherwise the result is that multiple
+ members can have the same name, since the variable was not reset
+ on each iteration of the loop. ........
+
+2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c: Add support for generating a ringing
+ sound on an incoming call. This is a bit of a hack. It just asks
+ the core to generate the same tone that it would when you hear
+ ringback when making an outbound call. But hey, it works, and you
+ get the localized ring tone for the appropriate language set on
+ the channel.
+
+ * channels/chan_console.c: Note that this module doesn't actually
+ play a ringing sound for an incoming call ... oops
+
+ * channels/chan_console.c: Show the correct CLI command to answer
+ the call
+
+2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: actually parse and store echocan parameters
+ from zapata.conf... this *should* work <G>
+
+2008-01-02 22:40 +0000 [r96071] Joshua Colp <jcolp@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't
+ use AST_C_DEFINE_CHECK for the two pthread things that may not
+ actually be definitions, they could be enums for example.
+
+2008-01-02 22:29 +0000 [r96028] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Add curly braces around a compound if
+ statement so that trunk will build properly
+
+2008-01-02 22:15 +0000 [r96021-96025] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 96024 via svnmerge ........ r96024 | russell
+ | 2008-01-02 16:14:28 -0600 (Wed, 02 Jan 2008) | 2 lines Convert
+ locks of the contexts list in pbx_config to the appropriate
+ rdlock or wrlock ........
+
+ * /: Blocked revisions 96022 via svnmerge ........ r96022 | russell
+ | 2008-01-02 16:04:47 -0600 (Wed, 02 Jan 2008) | 2 lines
+ pbx_dundi only needs a rdlock on the contexts list. ........
+
+ * /: Blocked revisions 96020 via svnmerge ........ r96020 | russell
+ | 2008-01-02 16:00:21 -0600 (Wed, 02 Jan 2008) | 2 lines
+ app_macro only needs a rdlock on the contexts list. ........
+
+2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: another
+ checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS
+ ioctl if it is present, but doesn't parse any supplied parameters
+ yet (this implementation is not very memory efficient as the
+ parameters and their values will be duplicated for each channel
+ that has the same settings, but we can worry about that later
+ once it is working)
+
+2008-01-02 21:49 +0000 [r96018] Russell Bryant <russell@digium.com>
+
+ * main/libresample/include/libresample.h: Add doxygen documentation
+ to libresample.h while it's still fresh on my mind
+
+2008-01-02 21:08 +0000 [r95994] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c,
+ apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo,
+ bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue
+ #11668, reported and patched by mvanbaak)
+
+2008-01-02 20:26 +0000 [r95947] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4
+ lines Allocate a SIP refer structure when performing a transfer
+ using BYE with Also so that the transfer information is properly
+ stored. (AST-2008-001) (closes issue #11637) Reported by:
+ greyvoip ........
+
+2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Since ',' is the standard argument separator in
+ trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of
+ AST_NONSTANDARD_APP_ARGS for determining member data.
+
+ * include/asterisk/app.h: Fix a typo in a comment.
+ AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'.
+
+2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: clean up hwgain CLI command and improve docs
+ for swgain CLI command
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate
+ the macro being checked for, but just check for its existence
+ finish implementation of check for Zaptel HWGAIN support add
+ check for Zaptel ECHOCANCEL_PARAMS support
+
+ * codecs/Makefile, include/asterisk/libresample.h (added),
+ codecs/codec_resample.c: and now just to keep the libresample
+ party going... if the functions from libresample are going to be
+ in the main Asterisk binary, it makes sense for the header that
+ defines them to be available without any special CFLAGS and to
+ out-of-tree modules building against /usr/include/asterisk
+
+ * channels/chan_zap.c: umm... this did not compile on x86-64, and
+ could not possibly have worked on any platform as it was passing
+ string pointers to a function expecting ints
+
+2008-01-02 18:05 +0000 [r95891] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan
+ 2008) | 9 lines A change to improve the accuracy of queue logging
+ in the case where a member does not answer during the specified
+ timeout period. Prior to this change, there was a small chance
+ that the member name recorded in this case would be blank. Also
+ prior to this change, if using the ringall strategy, if no one
+ answered the call during the specified timeout, the member name
+ listed in the queue log would randomly be one of the members that
+ was rung. (closes issue #11498, reported and tested by hloubser,
+ patched by me) ........
+
+2008-01-02 17:38 +0000 [r95888] Jason Parker <jparker@digium.com>
+
+ * apps/app_osplookup.c: Update osplookup documentation to use
+ commas instead of pipes. Closes issue #11666, patch by Laureano.
+
+2008-01-02 16:20 +0000 [r95864] Russell Bryant <russell@digium.com>
+
+ * main/Makefile, main/translate.c: For some odd reason, the last
+ set of libresample build changes from Kevin did not work for
+ everyone, but it did for some. This set of changes makes trunk
+ start again for those having problems. Instead of building
+ libresample as a static library, it just links the object files
+ in directly with the asterisk binary.
+
+2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile: fix some long-time breakage that kept
+ chan_misdn from being embedded
+
+ * channels/Makefile: use the proper technique for including
+ submodules so that embedding will work
+
+ * CHANGES: note that chan_console requires portaudio v19
+
+ * configure, configure.ac: actually check for a function present in
+ libiconv (don't know how this test could have worked before) and
+ don't do the check on Linux/GNU systems because libiconv is not
+ present there and attempting to link with '-liconv' always fails
+ (it's not necessary as the iconv functionality is always
+ available)
+
+ * main/libresample/src/filterkit.h,
+ main/libresample/src/resample.c,
+ main/libresample/win/libresample.dsp, main/libresample/configure,
+ main/libresample/Makefile.in, res/Makefile,
+ main/libresample/configure.in, main/libresample/src,
+ main/libresample/tests/testresample.c,
+ main/libresample/win/libresample.vcproj,
+ main/libresample/tests/compareresample.c, main/libresample/tests,
+ codecs/codec_resample.c, res/res_resample.c (removed),
+ main/libresample/README.txt, main/libresample/src/resamplesubs.c,
+ main/libresample/tests/resample-sndfile.c,
+ main/libresample/src/configtemplate.h,
+ main/libresample/install-sh, main/Makefile, main/translate.c,
+ main/libresample/include, main/libresample/src/resample_defs.h,
+ codecs/Makefile, main/libresample/config.guess,
+ main/libresample/config.sub, main/libresample/win,
+ main/libresample/LICENSE.txt, main/libresample (added),
+ main/libresample/Makefile.asterisk, build_tools/strip_nonapi,
+ res/libresample (removed), main/libresample/src/filterkit.c,
+ main/libresample/include/libresample.h: go back to including
+ libresample in the main Asterisk binary, but this time including
+ a small hack to ensure that it does get linked in (and also
+ modify the strip_nonapi script to leave the resample_<foo>
+ symbols alone)
+
+2008-01-02 11:34 +0000 [r95794] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Set stream flags to zero upon initialization.
+ When the XMPP over TLS/SSL connection resets for some reason, it
+ is wrongly believed as being secured, which makes the
+ re-connection process endlessly fail. This was reported by
+ mvanbaak in issue #11644.
+
+2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo <rizzo@icir.org>
+
+ * main/loader.c: some cleanup of this code while I am trying to
+ debug a problem with gdb dying while debugging asterisk. The
+ problem seems to be related with a race in the handling of
+ module_list, which in turn is triggeded by calling dlopen() on a
+ system which uses initializers to create locks.
+
+ * include/asterisk/module.h: There are three instances of the
+ module definition macros, which make maintaining this file very
+ error prone. This commit merges the embedded and !embedded
+ versions, and fixes the C++ version. Eventually we should move to
+ a single version of the macro. Too bad C++ doesn't like the
+ C-style struct initializers .foo = some_value
+
+2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant <russell@digium.com>
+
+ * res/libresample/src/resample_defs.h,
+ res/libresample/src/resample.c: Don't make libresample print out
+ debugging output
+
+ * main/translate.c: Make the translation table show slin16
+
+ * apps/app_meetme.c: fix a spacing issue introduced in revision
+ 95443.
+
+ * main/Makefile, res/libresample/README.txt, res/Makefile,
+ res/libresample/install-sh, res/libresample/configure,
+ res/libresample/Makefile.in, res/libresample/include,
+ codecs/Makefile, res/libresample/configure.in,
+ res/libresample/src, res/libresample/config.guess,
+ main/libresample (removed), res/libresample/config.sub,
+ res/libresample/win, codecs/codec_resample.c,
+ res/libresample/LICENSE.txt, res/libresample (added),
+ res/libresample/Makefile.asterisk, res/libresample/tests,
+ res/res_resample.c (added): Instead of linking libresample into
+ the main Asterisk binary, build it as res_resample, and mark
+ codec_resample as dependent upon res_resample. This prevents the
+ linker from optimizing away libresample, and also makes it so the
+ libresample code isn't linked in to multiple places. (I have
+ another module in a branch that needs it, too.)
+
+2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c: call directly the cli command to
+ implement hangup.
+
+ * channels/vcodecs.c: prevent a panic when destroying a channel
+ with no incoming video.
+
+ * channels/console_video.c: remove a leftover sleep(1) used for
+ debugging
+
+2008-01-01 23:09 +0000 [r95648] Joshua Colp <jcolp@digium.com>
+
+ * codecs/Makefile: Fix building of codec_resample on platforms
+ other then Cygwin. On everything else it actually gets built
+ after codec_resample, so you can't exactly link it in since it
+ doesn't exist.
+
+2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo <rizzo@icir.org>
+
+ * codecs/Makefile, codecs/codec_resample.c: make codec_resample
+ build on __CYGWIN__, and make it load on FreeBSD (and probably
+ other systems as well). Both need libresample.a to be specified
+ in the linking phase, and cygwin needs <float.h> as other BSD.
+ The checks for OS-specific headers should really be moved to some
+ common header though.
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ funcs/func_iconv.c, makeopts.in: implement "configure" checks for
+ libiconv, and add the iconv dependency for func_iconv. This fixes
+ some build issues on CYGWIN and FreeBSD and probably other
+ platforms where libiconv is not there by default
+
+2007-12-31 23:44 +0000 [r95578] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 95577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec
+ 2007) | 9 lines Avoiding a potentially bad locking situation.
+ ast_merge_contexts_and_delete writelocks the conlock, then calls
+ ast_hint_extension, which attempts to readlock the same lock.
+ Recursion with read-write locks is dangerous, so the inner lock
+ needs to be removed. I did this by copying the "guts" of
+ ast_hint_extension into ast_merge_contexts_and_delete (sans the
+ extra lock). (this change is inspired by the locking problems
+ seen in issue #11080, but I have no idea if this is the
+ problematic area experienced by the reporters of that issue)
+ ........
+
+2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant <russell@digium.com>
+
+ * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD.
+ Also, add some other platforms as they are likely the same.
+
+ * channels/chan_console.c: Update chan_console to natively use a 16
+ kHz sample rate. If it is talking to an 8 kHz endpoint, then
+ codec_resample will automatically be used to properly resample
+ the audio before sending it to/from chan_console.
+
+ * main/libresample/src/filterkit.h, main/libresample/README.txt,
+ main/libresample/tests/resample-sndfile.c,
+ main/libresample/src/resamplesubs.c, main/Makefile,
+ main/libresample/install-sh,
+ main/libresample/src/configtemplate.h,
+ main/libresample/src/resample.c,
+ main/libresample/win/libresample.dsp, main/libresample/configure,
+ main/libresample/Makefile.in, main/libresample/include, CHANGES,
+ main/libresample/src/resample_defs.h,
+ main/libresample/configure.in, main/libresample/src,
+ main/libresample/config.guess, codecs/Makefile,
+ main/libresample/tests/testresample.c, codecs/slin_resample_ex.h
+ (added), main/libresample/config.sub, main/libresample/win,
+ main/libresample/win/libresample.vcproj,
+ main/libresample/LICENSE.txt, main/libresample (added),
+ main/libresample/Makefile.asterisk, main/libresample/tests,
+ main/libresample/tests/compareresample.c, codecs/codec_resample.c
+ (added), main/libresample/src/filterkit.c,
+ main/libresample/include/libresample.h: Merge changes from
+ team/russell/codec_resample This commit imports libresample for
+ use in Asterisk. It also adds a new codec module, codec_resample.
+ This module uses libresample to re-sample signed linear audio
+ between 8 kHz and 16 kHz. It also provides an alternative for
+ converting between 16 kHz G.722 and 8 kHz signed linear when
+ using G.722, which will likely be useful as some people have
+ complained about volume issues when the current codec_g722
+ converts to 8 kHz signed linear. But, to test this, you will have
+ to disable the g722-to-slin and g722-to-slin16 translators in
+ codec_g722.c.
+
+2007-12-31 20:33 +0000 [r95490] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007)
+ | 3 lines Allow the default "0" to be returned if the STAT fails
+ (Closes issue #11659) ........
+
+2007-12-31 18:46 +0000 [r95443] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c: Fix a compiler warning (closes issue #11658,
+ reported and patched by eliel)
+
+2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant <russell@digium.com>
+
+ * configs/console.conf.sample (added), configs/modules.conf.sample,
+ channels/chan_console.c (added), CHANGES: Merge the main set of
+ changes from team/russell/chan_console. Add a new console channel
+ driver, chan_console, which is a console channel driver that uses
+ portaudio as a cross platform audio interface. It was written to
+ provide a console channel driver that works with Mac CoreAudio,
+ but it supports a number of other audio interfaces, as well,
+ including OSS and ALSA. It could one day be the single console
+ channel driver, but does not yet have as many features as
+ chan_oss.
+
+ * include/asterisk/channel.h: fix a spelling error in a comment
+
+ * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets
+ you set a config variable to a string field. (from
+ team/russell/chan_console)
+
+ * configure, include/asterisk/autoconfig.h.in: Regenerate configure
+ script to include check for portaudio.
+
+ * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add
+ configure script checking for portaudio.
+
+2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/vcodecs.c, channels/console_video.c, channels/Makefile,
+ channels/console_video.h, channels/vgrabbers.c (added): Move
+ grabbers definitions to a separate file, vgrabbers.c, so it is
+ easier to add more entries. This required moving struct grab_desc
+ to the common header, and adding an entry in the Makefile. On
+ passing, cleanup some comments and file headers (some are still
+ missing).
+
+ * channels/console_gui.c, channels/console_video.c: virtualize the
+ interface for video grabbers, which should make it easier to add
+ support for more grabbers (V4L2, firewire, and so on).
+
+ * channels/console_video.c: Add a few entries up to 1408x1152 in
+ the table of known video resolutions. This makes it very
+ convenient to enlarge images using the right-click on the video
+ window.
+
+ * channels/vcodecs.c, channels/console_video.c: change the
+ interface of video encapsulation routines, they only need the
+ buffer and mtu as input.
+
+ * channels/console_gui.c, channels/vcodecs.c,
+ channels/console_video.c, channels/console_video.h: various
+ rearrangements and renaming of console_video stuff
+
+2007-12-28 18:39 +0000 [r95233] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: The diff for this change looks really bad, but
+ all I did here was decrease the indentation of most of the
+ queue_exec function by reversing the logic of an if statement.
+ This change makes the function comply better with the coding
+ guidelines. Since this change is purely a cosmetic change to the
+ code, I am only committing the change to trunk.
+
+2007-12-28 18:26 +0000 [r95192] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) |
+ 6 lines Remove duplicate increment of the header count in the
+ add_header() function. (closes issue #11648) Reported by: makoto
+ Patch provided by sergee, committed patch by me, inspired by
+ comments from putnopvut ........
+
+2007-12-28 16:12 +0000 [r95167] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel
+ name is printed in verbose messages maximumWordLength option
+ added. Duration of words that do not meet the minimum word
+ duration will be logged The duration of pre-greeting silence will
+ be logged Only consider us in the greeting if we actually
+ detected a valid word duration. (closes issue #11650, reported
+ and patched by davevg)
+
+2007-12-28 08:57 +0000 [r95139] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_video.c: fix a small bug in printing out
+ geometries - wrong input.
+
+2007-12-28 00:17 +0000 [r95096] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec
+ 2007) | 8 lines I found a bug while browsing the queue code and
+ managed to reproduce it in a small setup. If a queue uses the
+ ringall strategy, it was possible through unfortunate coincidence
+ for a single member at a given penalty level to make app_queue
+ think that all members at that penalty level were unavailable and
+ cause the members at the next penalty level to be rung. With this
+ patch, we will only move to the next penalty level if ALL the
+ members at a given penalty level are unreachable. ........
+
+2007-12-27 23:32 +0000 [r95073] Luigi Rizzo <rizzo@icir.org>
+
+ * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove
+ more unnecessary casts for NULL. main/say.c is a big offender in
+ this respect.
+
+2007-12-27 23:28 +0000 [r95070] Jason Parker <jparker@digium.com>
+
+ * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and
+ document it as well. Closes issue #11645, patch by Laureano.
+
+2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo <rizzo@icir.org>
+
+ * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL
+ does not need to be cast to (char *)
+
+ * channels/chan_oss.c: remove useless casts
+
+2007-12-27 21:41 +0000 [r95025] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 95024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) |
+ 9 lines Don't report a syntax error when an empty string is
+ passed to ast_get_group. Just return 0. (closes issue #11540)
+ Reported by: tzafrir Patches: group_empty.diff uploaded by
+ tzafrir (license 46) -- slightly changed by me ........
+
+2007-12-27 20:11 +0000 [r94978] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/io.c: Merged revisions 94977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec
+ 2007) | 3 lines Fixing a typo in a comment. ........
+
+2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6
+ lines Include types.h in chan_h323 as without it it can not be
+ compiled on some operating systems like FreeBSD to name one.
+ (closes issue #11585) Reported by: sobomax Patches:
+ chan_h323.c.diff uploaded by sobomax (license 359) ........
+
+ * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4
+ lines Use ast_strlen_zero to see if our_contact is set or not on
+ the dialog. It is possible for it to be a pointer to NULL.
+ (closes issue #11557) Reported by: FuriousGeorge ........
+
+2007-12-27 17:26 +0000 [r94904] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/console_video.c: more
+ localization of gui stuff
+
+2007-12-27 17:18 +0000 [r94903] Mark Michelson <mmichelson@digium.com>
+
+ * doc/manager_1_1.txt: Adding documentation for new manager actions
+ and events in app_queue
+
+2007-12-27 16:51 +0000 [r94902] Luigi Rizzo <rizzo@icir.org>
+
+ * CHANGES: clarify the type of video support in chan_oss
+
+2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant <russell@digium.com>
+
+ * codecs/codec_g722.c: I went looking for where we downloaded the
+ g722 implementation and came across these two links. So, I'm
+ adding them so they are available for reference later.
+
+ * /: Blocked revisions 94831 via svnmerge ........ r94831 | russell
+ | 2007-12-27 09:16:56 -0600 (Thu, 27 Dec 2007) | 5 lines Now that
+ the contexts lock is a read/write lock, it should not be locked
+ here in ast_hint_state_changed(). This makes it get locked
+ recursively which now causes a deadlock. (closes issue #11080,
+ thanks to callguy for the access to a deadlocked machine)
+ ........
+
+ * /, main/translate.c, include/asterisk/translate.h: Merged
+ revisions 94828-94829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) |
+ 9 lines Change ast_translator_best_choice() to only pay attention
+ to audio formats. This fixes a problem where Asterisk claims that
+ a translation path can not be found for channels involving video.
+ (closes issue #11638) Reported by: cwhuang Tested by: cwhuang
+ Patch suggested by cwhuang, with some additional changes by me.
+ ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27
+ Dec 2007) | 2 lines Use the constant that I really meant to use
+ here ... ........
+
+2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson <oej@edvina.net>
+
+ * funcs/func_dialplan.c: This function checks more than just
+ contexts...
+
+ * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: -
+ This application needs better documentation and a RESULT code in
+ the dialplan.
+
+2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c, /: Merged revisions 94824 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007)
+ | 2 lines make this comment explain the situation in an even more
+ explicit fashion ........
+
+2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c: more steps to decouple the gui from the
+ rest of the code.
+
+ * channels/console_gui.c, channels/console_video.c,
+ channels/console_video.h: Enable building the code even if SDL is
+ not present (similarly, SDL is also detected at runtime). Now we
+ should be able to stream video even without a rendering device
+ (useful for remote monitoring).
+
+ * channels/console_gui.c, channels/console_video.c: more
+ localizations around sdl_setup
+
+ * channels/console_gui.c: use fread instead of mmap to read in the
+ comment area from the keypad. fread is simpler and more portable,
+ and there is no performance gain in using mmap.
+
+ * images/kpad2.jpg: update the region description with an empty
+ line at the beginning.
+
+2007-12-26 22:38 +0000 [r94818] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans
+ than 32. Also, rearrange compiler flags so the most often used
+ flags appear closer to the top. Reported by: tzafrir Patch by:
+ tzafrir,tilghman (Closes issue #11528)
+
+2007-12-26 22:29 +0000 [r94817] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/console_video.c: another bunch
+ of gui localizations
+
+2007-12-26 22:14 +0000 [r94814] Jason Parker <jparker@digium.com>
+
+ * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean
+ up the description and usage text a bit. Closes issue #11564,
+ patch by pnlarsson (with some extra cleanup by me).
+
+2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/console_video.c: more
+ localization of sdl stuff
+
+ * channels/console_gui.c, channels/console_video.c,
+ channels/console_video.h: move more gui stuff into console_gui.c
+
+2007-12-26 20:49 +0000 [r94809] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 94808 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007)
+ | 6 lines Workaround for what is probably a glibc bug (but we'll
+ see this crop up again and again, if we don't add the
+ workaround). Reported by: rolek Patch by: tilghman (Closes issue
+ #11601, closes issue #11426) ........
+
+2007-12-26 20:02 +0000 [r94806] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
+ apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c,
+ apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c,
+ pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c,
+ pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use
+ defined return values in load_module in more places. (closes
+ issue #11096) Patches: pbx_config.c.patch uploaded by moy
+ (license 222) pbx_dundi.c.patch uploaded by moy (license 222)
+ pbx_gtkconsole.c.patch uploaded by moy (license 222)
+ pbx_loopback.c.patch uploaded by moy (license 222)
+ pbx_realtime.c.patch uploaded by moy (license 222)
+ pbx_spool.c.patch uploaded by moy (license 222)
+ app_adsiprog.c.patch uploaded by moy (license 222)
+ app_alarmreceiver.c.patch uploaded by moy (license 222)
+ app_amd.c.patch uploaded by moy (license 222)
+ app_authenticate.c.patch uploaded by moy (license 222)
+ app_cdr.c.patch uploaded by moy (license 222)
+ app_zapateller.c.patch uploaded by moy (license 222)
+ app_zapbarge.c.patch uploaded by moy (license 222)
+ app_zapras.c.patch uploaded by moy (license 222)
+ app_zapscan.c.patch uploaded by moy (license 222)
+
+2007-12-26 20:01 +0000 [r94805] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c, channels/vcodecs.c,
+ channels/console_video.c, channels/console_video.h: more
+ preparation for untangling of the various console_video stuff
+
+2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 94801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) |
+ 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already
+ set before starting autoservice, remember it and ensure that the
+ channel has the same setting when autoservice gets stopped.
+ (pointed out by d1mas, patched up by me) ........
+
+ * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan
+ function, DIALPLAN_EXISTS(), which allows you to check for the
+ existence of a dialplan target. (closes issue #11579) Reported
+ by: irroot Patches: func_dialplan2.c uploaded by irroot (license
+ 52) -- Additional changes by me.
+
+ * main/autoservice.c, /: Merged revisions 94797 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) |
+ 4 lines When a channel is in autoservice, mark a flag on the
+ channel that says that we only care about the END of a digit.
+ That way, no magic digit emulation stuff will happen when all
+ we're doing is queueing up END frames. ........
+
+ * main/channel.c: Leave a note for a minor bug that was pointed out
+ by d1mas
+
+2007-12-26 18:05 +0000 [r94795] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: Convert raw bits for callprogress bitfield
+ to use constants, for greater code clarity Reported by: dimas
+ Patch by: dimas (Closes issue #11280)
+
+2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant <russell@digium.com>
+
+ * /, res/res_features.c: Merged revisions 94793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) |
+ 3 lines Don't try to send a parked call back to itself. (closes
+ issue #11622, reported by djrodman, patched by me) ........
+
+ * Makefile, /: Merged revisions 94789 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) |
+ 5 lines List include/asterisk/version.h as a .PHONY target
+ because we want the commands listed for this target to be
+ executed regardless of whether the file exists or not. This fixes
+ having the version not up to date when running from svn. (closes
+ issue #11619, reported by plack, fixed by me) ........
+
+ * main/autoservice.c, /: Merged revisions 94790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) |
+ 5 lines Don't store DTMF BEGIN frames while a channel is in
+ autoservice. It's just going to make ast_read() do a lot of extra
+ work when the channel comes back out of autoservice. (closes
+ issue #11628, patched by me) ........
+
+ * channels/chan_iax2.c: Fix a bug in peer handling that caused
+ multiple instances of a peer to end up in the peers container
+ after a reload. Somehow, this bug doesn't exist in 1.4 ...
+ (closes issue #11626) (reported by pnlarsson, additional info
+ from mvanbaak, fixed by me)
+
+ * utils: update svn:ignore for astcanary
+
+2007-12-26 15:58 +0000 [r94782] Mark Michelson <mmichelson@digium.com>
+
+ * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding
+ support for storing the queue log entries in a realtime backend.
+ (closes issue #11625, reported and patched by sergee) Thank you
+ very much to sergee for adding this new feature!
+
+2007-12-26 10:14 +0000 [r94774] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_gui.c (added), channels/vcodecs.c (added),
+ channels/console_video.c: Split console_video.c so that video
+ codecs and gui functions are in separate files (still #include'd
+ because of tangling in the data structures, but this is going to
+ be cleaned up). The video grabbing functions still need to be
+ moved to a separate file.
+
+2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_pickupchan.c (added): Add pickup by channel (Closes
+ issue #11161)
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: Change the
+ abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF
+ character. Also, fix the documentation to match the code.
+
+ * res/res_agi.c: Add channel thread ID to the information passed to
+ AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162)
+
+2007-12-25 02:28 +0000 [r94770] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 94769 via svnmerge ........ r94769 | file |
+ 2007-12-24 22:27:08 -0400 (Mon, 24 Dec 2007) | 2 lines file
+ says... build on the builders. ........
+
+2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 94767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007)
+ | 5 lines Race: we need to wait to queue a NewChannel event until
+ after the channel is inserted into the channel list. The reason
+ is because some manager users immediately queue requests from the
+ channel when they see that event and are confused when Asterisk
+ reports no such channel. (Closes issue #11632) ........
+
+ * /: Blocked revisions 94765 via svnmerge ........ r94765 |
+ tilghman | 2007-12-24 10:17:01 -0600 (Mon, 24 Dec 2007) | 5 lines
+ More deadlock avoidance code (this time between sip_monitor and
+ sip_hangup) Reported by: apsaras Patch by: tilghman (Closes issue
+ #11413) ........
+
+ * /: Merged revisions 94763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007)
+ | 5 lines Another bit of bad logic in realtime_peer Reported by:
+ dimas Patch by: dimas (Closes issue #11631) ........
+
+2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_video.c, channels/console_video.h: support
+ sdl_videodriver to send output to x11/aalib/console
+
+ * channels/console_video.c: move reading info from the keypad to a
+ separate function. Remove an unused keypad field and some
+ debugging messages. Adjust formatting on config file parsing
+
+ * channels/console_video.c: make sure the minimum surface depth is
+ 16bpp so we can create YUVoverlays. With this change we can do
+ setenv SDL_VIDEODRIVER aalib and output to an ascii window (which
+ is still in an X11 window). If you also do unsetenv DISPLAY then
+ the output goes into the main asterisk window, unfortunately it
+ interferes with the normal output so you don't see much. In any
+ case, i don't think we are very far away from having a working
+ xterm videophone!
+
+ * channels/Makefile: avoid rebuilding dependent files if the
+ generated busy.h and ringtone.h do not change. Ths masks (but
+ does not solve) a but that i am seeing in doing a 'gmake install'
+ without donig a 'gmake all' first.
+
+2007-12-23 01:38 +0000 [r94662] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007)
+ | 2 lines Argh... I suppose third time's the charm. ........
+
+2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo <rizzo@icir.org>
+
+ * configs/oss.conf.sample, channels/console_video.c: Change the
+ name of config file entries for keypad regions from
+ 'keypad_entry' to 'region'. Fix the example file accordingly.
+ Also make some fixes in the code do reset entries on reload of
+ the keypad. The recently committed kpad2.jpg has the correct
+ names.
+
+ * images/kpad2.jpg (added): add a sample keypad (with annotations)
+ for console video
+
+ * channels/console_video.c, channels/Makefile, channels/chan_oss.c,
+ channels/console_video.h (added): Build console_video support by
+ linking in, as opposed to including, console_video.c This will
+ ease the task of splitting console_video.c into its components
+ (V4L and X11 grabbers, various video codecs and packetizers,
+ SDL), as well as ease future extensions (e.g. additional video
+ sources, codecs and rendering engines). For the time being
+ nothing changes for users: video support is off by default, and
+ requires -DHAVE_VIDEO_CONSOLE on the command line to be included
+ (if SDL and FFMPEG are available).
+
+2007-12-21 21:19 +0000 [r94593] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Something I've been itching to do for a
+ while now. A minor optimization in app_voicemail. Since the
+ dtable in base_encode always gets populated with the same values
+ every time and never changes, make it static and const and only
+ initialize it once. Also, there's no reason to define
+ BASEMAXINLINE twice, so remove the redundant #define.
+
+2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: We should only clear this value if we have
+ to
+
+ * channels/chan_zap.c: Commit non TCP transport part of #11506.
+ Includes numerous additional parameters, as well as RLT support
+ for DMS type switches
+
+2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_config_sqlite.c: Store dates using local time instead of
+ UTC (closes issue #11610, reported and patched by
+ rbraun_performatique)
+
+ * apps/app_queue.c: Fix a memory leak when reloading queue rules.
+
+ * CHANGES: The one documentation source I forgot to update after
+ the merge of the queue-penalty branch was the CHANGES file. No
+ longer!
+
+ * apps/app_voicemail.c: Lots of coding guidelines cleanup.
+
+ * /: Blocked revisions 94543 via svnmerge ........ r94543 |
+ mmichelson | 2007-12-21 14:21:59 -0600 (Fri, 21 Dec 2007) | 3
+ lines Bunch of coding guidelines cleanup ........
+
+ * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec
+ 2007) | 8 lines Better quota support for using IMAP storage
+ voicemail (closes issue #11415, reported by jaroth) (closes issue
+ #11152, reported by selsky) Patch provided by jaroth ........
+
+2007-12-21 20:12 +0000 [r94541] Jason Parker <jparker@digium.com>
+
+ * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
+ codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c,
+ codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c:
+ codecs.conf really shouldn't be mandatory.. it never had been
+ before, so let's go back to being optional. A big "thank you" to
+ pnlarsson on IRC for allowing me access to his system to debug
+ this. Closes issue #11584.
+
+2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec
+ 2007) | 5 lines The mail_copy c-client function does not expect a
+ full imap mailbox string, just the name of the mailbox. (closes
+ issue #11419, reported and patched by jaroth, with additional
+ patchwork from me) ........
+
+ * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in
+ trunk
+
+ * main/dial.c, /: Merged revisions 94468 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec
+ 2007) | 6 lines Since we are freeing list elements within a list
+ traversal, we need to use the safe traversal and remove the item
+ from the list before freeing it. (closes issue 11612, reported by
+ dtyoo) ........
+
+2007-12-21 16:42 +0000 [r94467] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 94466 via svnmerge ........ r94466 | russell
+ | 2007-12-21 10:37:47 -0600 (Fri, 21 Dec 2007) | 6 lines Convert
+ the contexts lock to a read/write lock to resolve a deadlock.
+ This has a nice side benefit of improving performance. :) (closes
+ issue #11609) (closes issue #11080) ........
+
+2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec
+ 2007) | 3 lines Removing a debug message I accidentally just
+ committed ........
+
+ * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec
+ 2007) | 5 lines Fixing Portuguese syntax for saying dates and
+ times. Also some coding guidelines cleanup. (closes issue #11599,
+ reported and patched by caio1982, coding guidelines cleanup by
+ me) ........
+
+2007-12-21 15:14 +0000 [r94419] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 94418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007)
+ | 2 lines Fix for restart-as-user problem reported via the -dev
+ list ........
+
+2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Moved the update of the queue_ent's rule list
+ to just before we try to call queue members. This allows for the
+ change in penalty levels to be executed at the most logical time
+ frame.
+
+ * configs/queues.conf.sample, doc/tex/channelvariables.tex,
+ apps/app_queue.c, configs/queuerules.conf.sample (added): Merging
+ the queue-penalty branch. In short, this allows one to
+ dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced
+ QUEUE_MIN_PENALTY during a call depending on the amount of time
+ passed. The purpose is to allow the call to open up to more (or
+ maybe just different) members without the caller's losing his
+ place in the queue. See configs/queuerules.conf.sample for an
+ example of how to set up queue rules and
+ configs/queues.conf.sample for how to associate a rule with a
+ queue. Along with the functional changes, new CLI and manager
+ commands exist to show the rules defined and there is an
+ additional CLI command to reload the queue rules. Future
+ enhancements that may be made: support for realtime queue rules
+ and support for dynamically adding a rule through the manager or
+ CLI. Also a manager command to reload the queue rules (I'll
+ probably write this myself very soon).
+
+ * apps/app_voicemail.c: The changes to header inclusion in trunk
+ broke compilation of app_voicemail when using IMAP storage. The
+ reason is that c-client has its own definitions for LOG_WARNING
+ and LOG_DEBUG, so we need to be sure to include asterisk's
+ definitions last so that we use the proper values in
+ app_voicemail. (closes issue #11437, reported by blitzrage, patch
+ suggested by blitzrage)
+
+2007-12-20 22:39 +0000 [r94320] Russell Bryant <russell@digium.com>
+
+ * configs/zapata.conf.sample: Add a bit more to the description of
+ the "mwimonitor" option.
+
+2007-12-20 22:28 +0000 [r94319] Steve Murphy <murf@digium.com>
+
+ * build_tools/make_buildopts_h: closes issue #11287; thanks to
+ snuffy for this fix, which will surely make all solaris owners
+ shout praises to his name.
+
+2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r94256 | russell | 2007-12-20 14:22:22 -0600
+ (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20
+ Dec 2007) | 5 lines Fix another potential seg fault ... (closes
+ issue #11606) Reported by: dimas ........ ................
+
+ * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) |
+ 10 lines Fix a deadlock in d-channel handling in chan_zap. This
+ deadlock was introduced by the fix to ensure that channels are
+ properly locked when handling channel variables. There were
+ sections of this code where the channel pvt was locked before the
+ channel lock, when in fact it _must_ be the other way around.
+ (closes issue #11582) Reported by: bugi ........
+
+2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_usbradio.c, include/asterisk/config.h,
+ channels/console_video.c, channels/chan_oss.c: add some macros to
+ simplify parsing the config file, see description in config.h .
+ They are a variant of the set of macros i used in chan_oss.c,
+ structured in a way to be more robust to the presence of spurious
+ ';' - basically, they define wrappers for 'do {' and '} while
+ (0)', plus some helper functions to deal with simple cases such
+ as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix
+ (CV_ as 'Config Variable') tries to be easy to remember and has
+ been chosen to not conflict with other existing macros in the
+ tree. For the time being, I have only updated the three source
+ files in the tree that used the old M_* macros. Hopefully, more
+ files will be converted. NOTE: I understand that inventing my own
+ dialect of C is generally wrong; however, the lack of adequate
+ support in the language encourages lazy programming practices
+ (such as ignoring errors, bounds, etc.) and this increases the
+ chance of vulnerability in the code, especially because we are
+ parsing user input here. Hopefully, these macros and the use of
+ ast_parse_arg (in config.h) should encourage the programmer to
+ write more robust code.
+
+ * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c,
+ utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify
+ http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that
+ paths and filename are writable by asterisk.c without causing
+ segfaults. This involves defining the variables as const char *,
+ and having them point to as static, writable buffer defined in
+ asterisk.c On passing, fix some errors in using these variables
+ in some files in utils/ , and in res/snmp/agent.c which was
+ redefining a variable without using paths.h (not applicable to
+ 1.4)
+
+2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the
+ set_member_penalty code. 2. Fix an error when checking the CLI
+ command for setting a member's penalty. 3. Fix a logging error if
+ the incorrect parameter was the queue name or interface. (closes
+ issue #11544, reported and patched by Laureano)
+
+ * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec
+ 2007) | 6 lines Sox versions 13.0.0 and newer do not have
+ "soxmix" and instead use sox -m. res_monitor needs to use this if
+ the user does not have soxmix. (closes issue #11589, reported by
+ amessina, patch inspired by amessina but with a flourish from me)
+ ........
+
+2007-12-19 22:51 +0000 [r94085] Russell Bryant <russell@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 94077 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) |
+ 4 lines Check for the existence of the soxmix application on the
+ target platform and have the result available in autoconfig.h.
+ (part of issue #11589) ........
+
+2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Add 'voicemail reload' command. Reported
+ by: eliel Patch by: eliel (Closes issue #11365)
+
+ * apps/app_waituntil.c (added): Add contributed WaitUntil app.
+ Original code by pprindeville, updated for trunk by tilghman.
+ (Closes issue #11487)
+
+2007-12-19 19:29 +0000 [r94029] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/time.h: Add a couple of new time API calls -
+ ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported
+ by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license
+ 88)
+
+2007-12-19 17:58 +0000 [r94002] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/console_video.c: Add instructions on how to generate
+ your own font.
+
+2007-12-19 17:31 +0000 [r93956] Joshua Colp <jcolp@digium.com>
+
+ * /: Merged revisions 93955 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2
+ lines Make the 1.4 builders happy, ensure var is NULL. ........
+
+2007-12-19 17:13 +0000 [r93952] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007)
+ | 3 lines Avoid segfault in chan_iax when peer isn't defined
+ (Closes issue #11602) ........
+
+2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo <rizzo@icir.org>
+
+ * main/utils.c, include/asterisk/strings.h: Add a new API function,
+ written at least twice in app_voicemail.c and likely in other
+ places too. This is quite useful when placing mail/html stuff in
+ config files. /*! \brief Convert some C escape sequences
+ (\b\f\n\r\t) into the equivalent characters. \brief s The string
+ to be converted (will be modified). \return The converted string.
+ */ char *ast_unescape_c(char *s);
+
+ * include/asterisk/config.h, main/config.c: add support for
+ PARSE_DOUBLE, and remove identifiers for types not supported
+ (INT16 and UINT16)
+
+2007-12-19 09:20 +0000 [r93899] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too
+ large...
+
+2007-12-19 08:57 +0000 [r93898] Luigi Rizzo <rizzo@icir.org>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK()
+ and use it to check for NETSMP and GTK (GTK is not used thoug).
+ AST_EXT_TOOL_CHECK() could be used for checking curl status as
+ well, perhaps with a small addition because we currently seem to
+ require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE
+ entry in makeopts.in We don't have yet any macros for using
+ pkg-config to check for a specific package (right now there is
+ only gtk2+ in the category).
+
+2007-12-19 08:57 +0000 [r93897] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the
+ ability to specify the To: header in an outbound INVITE by adding
+ an exclamation mark to the dial string. This patch also exists
+ for 1.4 in the fixtoheader-1.4 branch and has been in production
+ for quite some time.
+
+2007-12-19 08:12 +0000 [r93875] Luigi Rizzo <rizzo@icir.org>
+
+ * res/snmp/agent.c: make netsmp build under AST_DEVMODE.
+ Description, included in the source, is below. I should note that
+ the PACKAGE_* macros that asterisk defines in autoconfig.h are
+ not used anywhere in the tree so they should just be removed. /*
+ * There is some collision collision between netsmp and asterisk
+ names, * causing build under AST_DEVMODE to fail. * * The
+ following PACKAGE_* macros are one place. * Also netsnmp has an
+ improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H
+ instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define
+ it to 0, however this will fail * when the proper check is
+ implemented. */ No
+
+2007-12-19 07:01 +0000 [r93854] Olle Johansson <oej@edvina.net>
+
+ * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for
+ starting remote Asterisk by naming the actual runtime socket
+ instead of pointing to configuration file with -C Reported by:
+ sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax
+ (license 359) doc changes by committer (closes issue #11598)
+
+2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * apps/app_osplookup.c: add missing header file
+
+2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Making the canary error message a little more
+ obvious.
+
+ * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a
+ canary process, for high priority mode (asterisk -p) to ensure
+ that if Asterisk goes into a busy loop, the machine will be
+ recoverable. We'd still need to do a restart to put Asterisk back
+ into high priority mode, but at least a reboot won't be required.
+ (Closes issue #11559)
+
+2007-12-18 22:44 +0000 [r93765] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 93764 via svnmerge ........ r93764 | qwell |
+ 2007-12-18 16:42:41 -0600 (Tue, 18 Dec 2007) | 4 lines FreeBSD
+ also does not have byte swap functions. Issue 11586, patch by
+ sobomax. ........
+
+2007-12-18 21:13 +0000 [r93741] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Move some warnings away to debug since some
+ devices send a packet with a silly string as a NAT keepalive
+ packet.
+
+2007-12-18 18:39 +0000 [r93672] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 93668 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600
+ (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18
+ Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119)
+ ........ ................
+
+2007-12-18 18:20 +0000 [r93666] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/paths.h: remove a leftover line with only a '#'
+ (wonder why the compiler does not complain!) and variables that
+ are only used in asterisk.c
+
+2007-12-18 17:05 +0000 [r93626] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 93625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec
+ 2007) | 6 lines Rework deadlock avoidance used in ast_write,
+ since it meant that agent channels which were being monitored had
+ one audio file recorded and one empty audio file saved. (closes
+ issue #11529, reported by atis patched by me) ........
+
+2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c,
+ utils/ael_main.c, utils/extconf.c, main/asterisk.c,
+ res/res_monitor.c, utils/conf2ael.c: make configuration variable
+ const so they are not accidentally modified. This requires
+ casting the strings in asterisk.c when writing to them, so we do
+ it through a macro to do it consistently.
+
+ * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c,
+ apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c,
+ main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c,
+ cdr/cdr_csv.c: remove unnecessary (char *) casts for
+ ast_config_AST_* variables. There are some left in the .flex
+ files, left to the maintainer...
+
+ * build_tools/make_defaults_h, main/asterisk.c: Rename the macros
+ in defaults.h - they are not meant to be globally visible.
+ Document the fact that DEFAULT_TMP_DIR cannot be overridden from
+ the default configuration (this needs to be fixed, as you could
+ have a totally different spooldir configured at runtime, and yet
+ DEFAULT_TMP_DIR keeps the compile-time default). Remove two
+ unused entries for sounds and images.
+
+ * Makefile.moddir_rules: make the code match documentation - now
+ you can specify multiple words in MODULE_PREFIX.
+
+ * CREDITS: Name the people responsible for some recent
+ contributions to the tree.
+
+ * Makefile: Two small changes: + document the difference between
+ "A=foo make ..." and "make A=foo ..." and suggest using
+ COPTS/LDOPTS if you need to use the second form to pass compiler
+ and loader flags; + define only in one place the environment used
+ to build stuff in menuselect/
+
+2007-12-18 07:56 +0000 [r93557] Olle Johansson <oej@edvina.net>
+
+ * doc/CODING-GUIDELINES: A minor update, caused by a recent bug
+ report ;-)
+
+2007-12-18 07:22 +0000 [r93536] Luigi Rizzo <rizzo@icir.org>
+
+ * doc/CODING-GUIDELINES: small documentation update (nothing
+ important).
+
+2007-12-18 02:57 +0000 [r93514] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_unistim.c: You... will... build! I say so and
+ therefore you will.
+
+2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor
+ cleanups
+
+2007-12-17 23:10 +0000 [r93464] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_unistim.c: fix building under cygwin. At this point
+ WINARCH should go away.
+
+2007-12-17 22:57 +0000 [r93424] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 93420 via svnmerge ........ r93420 | qwell |
+ 2007-12-17 16:56:58 -0600 (Mon, 17 Dec 2007) | 1 line Missed a
+ spot.. ........
+
+2007-12-17 22:54 +0000 [r93405] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_unistim.c: remove some unnecessary includes
+
+2007-12-17 22:50 +0000 [r93390] Jason Parker <jparker@digium.com>
+
+ * /, main/translate.c: Merged revisions 93381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4
+ lines What was I thinking when I wrote this masterpiece? -1 + 1 =
+ 0.. who woulda thunk it?. ........
+
+2007-12-17 22:38 +0000 [r93380] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_oss.c: surprising as it may be, chan_oss compiles
+ correctly under cygwin as well, provided you look for soundcard.h
+ in the right place...
+
+2007-12-17 22:29 +0000 [r93378] Joshua Colp <jcolp@digium.com>
+
+ * /, main/utils.c: Merged revisions 93377 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7
+ lines Do not try to access information about a lock when printing
+ out a trylock attempt. It is possible for the lock that it
+ references to no longer be valid. This would have caused
+ segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
+ Reported by: callguy (closes issue #11100) Reported by: callguy
+ ........
+
+2007-12-17 21:14 +0000 [r93337] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17
+ Dec 2007) | 6 lines Today is tomorrow's yesterday, and
+ yesterday's tomorrow is today, and tomorrow's tomorrow is the day
+ after tomorrow, so who cares if you recycle anyway? If this
+ confuses you, that's nothing compared to what this fixes. ;-)
+ ........
+
+2007-12-17 21:12 +0000 [r93335] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c,
+ channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions
+ 93182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8
+ lines Issue 11574: Add dependencies on res_monitor and
+ res_features. I wonder if Asterisk can run at all without
+ res_features. My guess is that there's propably a lot of more
+ modules and the core that depends on it. Reported by: caio1982
+ (closes issue #11574) ........
+
+2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Removing some leftover debug messages from a
+ while back.
+
+ * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec
+ 2007) | 6 lines We need to create the directory for a voicemail
+ user even if they are using IMAP storage since greetings are
+ stored in the filesystem. (closes issue #11388, reported by
+ spditner, patch by me inspired by a patch by spditner) ........
+
+2007-12-17 18:07 +0000 [r93252] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6
+ lines If a call is received with a called number IE containing
+ nothing go to the 's' extension. (closes issue #9099) Reported
+ by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by
+ Corydon76 (license 14) ........
+
+2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils: all created files need to be listed in the ignore property
+
+ * channels/chan_unistim.c, build_tools/menuselect-deps.in,
+ configure, configure.ac, channels/Makefile, channels/chan_oss.c:
+ make the configure script detect that it is running on a Windows
+ platform, and report that information so that menuselect can use
+ it (all information that is used to decide whether to build
+ modules or not must be fed to menuselect so the user knows what
+ will be built and why... don't make module build decisions in the
+ makefiles, please)
+
+ * Makefile: make using PRINT_DIR a little easier
+
+2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix usage of rtptimeout. It can be used
+ without rtpkeepalive, and the value can not be accessed directly
+ in the SIP pvt structure. All RTP related timeouts have to be
+ retrieved using the ast_rtp_* function calls. (closes issue
+ #11562) Reported by: ibc
+
+ * channels/chan_unistim.c: If no timezone is available use the
+ default message. (closes issue #11576) Reported by: junky
+
+ * channels/chan_unistim.c: Make chan_unistim actually be able to
+ unload. When creating a thread that you want to pthread_join you
+ have to explicitly create it as joinable, and also if using
+ pthread_cancel you have to have a pthread_testcancel to see if it
+ has been called.
+
+2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Blocked revisions 93183 via svnmerge ........ r93183 |
+ kpfleming | 2007-12-16 23:21:08 -0800 (Sun, 16 Dec 2007) | 2
+ lines fix some copy-and-paste leftovers ........
+
+ * codecs, /, build_tools/make_version,
+ include/asterisk/autoconfig.h.in, configure.ac, apps,
+ Makefile.moddir_rules, res/Makefile, pbx/Makefile,
+ build_tools/prep_moduledeps (removed), channels/Makefile, cdr,
+ formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
+ configure, build_tools/embed_modules.xml, cdr/Makefile,
+ build_tools/prep_tarball, makeopts.in, formats/Makefile, res,
+ pbx, channels, funcs/Makefile: Merged revisions 93180 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007)
+ | 23 lines In
+ http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
+ rizzo brought up some issues related to the way that the metadata
+ required for menuselect and the rest of the build system is
+ extracted from the source files. Since I had a few hours to kill
+ on an airplane today, I decided to improve this situation... so
+ now the system caches the extracted metadata and uses it to build
+ the menuselect 'tree' as much as it can. The result of this is
+ that when a single source file is changed, only the metadata for
+ that file needs to be extracted again, and the rest is used from
+ the cache files. I also reduced the number of forked processes
+ required to do the metadata extraction; it was actually possible
+ to do most of what we needed in the Makefiles themselves without
+ using any shell scripts at all! On my laptop, these changes
+ resulted in an 80% decrease in the time required for the
+ 'menuselect.makeopts' automatic check to occur after editing a
+ single source file. While doing this work I also cleaned up a few
+ minor things in the Makefiles, adding a check for 'awk' to the
+ configure script and changed all remaining places we use 'grep'
+ or 'awk' to use the ones found by the configure script, and
+ changed the 'prep_tarball' script to build the menuselect
+ metadata so that tarballs of Asterisk will include it and won't
+ require the user to wait while it is extracted after unpacking.
+ ........
+
+2007-12-16 19:06 +0000 [r93173] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: menuselect.makeopts is not a .PHONY target
+
+2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson <oej@edvina.net>
+
+ * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks,
+ dimas! (closes issue #11572) Reported by: dimas Patches:
+ dundilog-trunk.patch uploaded by dimas (license 88)
+
+ * main/manager.c, CHANGES: Adding a new CLI command for "manager
+ reload", which is important now that you need to reload after
+ changes. Thanks YS. Reported by: ys Patches:
+ trunk93163_manager_reload.c.diff uploaded by ys (license 281)
+ (related to issue #11414)
+
+ * main/manager.c, CHANGES: Change manager so that registered
+ accounts are stored in memory. This opens for a manager realtime
+ implementation. If you change accounts in manager.conf, you now
+ need to reload to activate the changes (deletions, additions).
+ This was not the case with 1.4. Reported by: ys Patches:
+ trunk93163_manager_reload.c.diff uploaded by ys (license 281)
+ (closes issue #11414)
+
+ * CHANGES: Adding console_video to CHANGES. It's important that we
+ keep this file up to date, even with experimental stuff.
+
+ * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample,
+ channels/chan_sip.c, include/asterisk/rtp.h,
+ include/asterisk/netsock.h, channels/iax2-provision.c,
+ UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample,
+ CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c,
+ configs/h323.conf.sample, configs/iax.conf.sample,
+ channels/chan_skinny.c, configs/mgcp.conf.sample,
+ configs/unistim.conf.sample, channels/chan_h323.c,
+ configs/iaxprov.conf.sample, pbx/pbx_dundi.c,
+ configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements
+ to QoS/CoS handling by IgorG - Refer to the proper documentation
+ - Implement separate signalling/media QoS/CoS in many channels
+ using RTP - Improve warnings and verbose messages - Deprecate
+ some old settings Minor modifications by me, a big effort from
+ IgorG. Thanks! Reported by: IgorG Patches:
+ qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
+ Tested by: IgorG (closes issue #11145)
+
+2007-12-16 10:34 +0000 [r93162] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: use a simpler idiom for 'cmp -s ...'
+
+2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson <oej@edvina.net>
+
+ * main/asterisk.c: Don't drop the first character randomly in long
+ listings in the CLI. Reported by: slavon Patches:
+ asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich
+ (license 176) Tested by: eliel (closes issue #9325)
+
+ * configs/sip.conf.sample, CHANGES: Update documentation
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Make more timers
+ settable in SIP so that we can force timeout earlier on
+ non-responsive SIP servers. Thanks, jcmoore, for the patch!
+ Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt
+ uploaded by jcmoore (license 9) (closes issue #9771)
+
+ * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo
+ after all. Didn't a clever guy once say "Compile before you
+ commit" ? :-)
+
+2007-12-15 08:10 +0000 [r93151] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/file.h: fix a typo from revision 93138
+
+2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo <rizzo@icir.org>
+
+ * configs/oss.conf.sample: configuration options related to video
+ support.
+
+ * channels/console_video.c (added): Bring in video console support
+ for chan_oss (and later chan_alsa too). This is disabled in the
+ default build, you need to explicitly enable it compiling with
+ make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do
+ a video call with chan_oss, using the webcam (or X11 grabbing) as
+ local source, and rendering the incoming stream on your screen.
+ Currently supported formats are h261, h263, h263+, h264, mpeg4
+ (all through the avcodec lib, part of ffmpeg). Incoming video is
+ on the left, outgoing video is on the right, while the center
+ displays a keypad (if configured so). Right clicking on the video
+ windows increases the size, center clicking reduces the size.
+ Dragging the mouse (with the left key) on the right window while
+ the X11 grabber is active moves the grab area. This is the result
+ of work by Sergio Fadda, Marta Carbone and myself, all properly
+ disclaimed to digium. Note, there is a lot of work left to do in
+ this module, including adding support for Video4LinuxV2 (I have
+ patches from Matteo Brancaleoni which should be integrated), and
+ making the GUI a lot more friendly than it is now (e.g.
+ supporting merging or switching among multiple sources, a text
+ window, and more).
+
+ * channels/chan_oss.c: remove some redundant headers
+
+ * include/asterisk/file.h: include mmap header if detected by
+ configure
+
+2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Resolve a compiler warning
+
+ * apps/app_voicemail.c: Change places where the name "INBOX" was
+ hardcoded to use the imapfolder setting from voicemail.conf
+ instead. This commit will help to get issue #11415 moving towards
+ commitment.
+
+2007-12-14 21:09 +0000 [r93090] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h:
+ Solaris compat fixes Reported by: snuffy Patch by:
+ snuffy,tilghman (Closes issue #11315)
+
+2007-12-14 19:31 +0000 [r93067] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: make something static
+
+2007-12-14 19:27 +0000 [r93066] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_privacy.c, UPGRADE.txt, CHANGES,
+ configs/privacy.conf.sample (removed): Remove use of privacy.conf
+ by the Privacy app. Reported by: eliel Patch by: eliel (Closes
+ issue #11344)
+
+2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to
+ increment the numbers used on the VERBOSITY_ATLEAST calls by 1.
+ Thanks to kpfleming for pointing this out.
+
+ * include/asterisk/logger.h, main/pbx.c, main/manager.c,
+ funcs/func_timeout.c: Changed VERBOSITY_LEVEL to
+ VERBOSITY_ATLEAST to be more accurate.
+
+ * include/asterisk/logger.h, main/pbx.c, main/manager.c,
+ funcs/func_timeout.c, main/logger.c: After reading Russell's
+ e-mail to the dev list stating that checking option_verbose is
+ not equivalent to the check done by ast_verb, I wrote a macro,
+ VERBOSITY_LEVEL, which does this check. I did a quick look in the
+ source and used this macro in some places where option_verbose
+ was used. I also converted some verbose messages in logger.c to
+ use ast_verb instead of ast_verbose.
+
+2007-12-14 18:24 +0000 [r93041] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: gcc 4.1.3 wants a union used here.
+
+2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant <russell@digium.com>
+
+ * main/config.c: Print an error message if a #included file does
+ not exist
+
+ * /: Blocked revisions 93000 via svnmerge ........ r93000 | russell
+ | 2007-12-14 11:36:08 -0600 (Fri, 14 Dec 2007) | 7 lines There
+ are a lot of existing systems that #include non-existent files.
+ So, to make the transition to treating this as an error a bit
+ less painless, just issue a huge error message for now. Then,
+ later, we can reinstate the code that treats it as a failure.
+ (Thanks to philippel for the feedback) ........
+
+2007-12-14 17:29 +0000 [r92999] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Publish the AGI events to manager. Reported by:
+ moy Patch by: moy,tilghman (Closes issue #11337)
+
+2007-12-14 15:59 +0000 [r92976] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_timeout.c: Reintroduce an optimization that was lost
+ when converting trunk to use ast_verb.
+
+2007-12-14 15:49 +0000 [r92939] Tilghman Lesher <tlesher@digium.com>
+
+ * main/editline/sys.h: If malloc.h is included in a Solaris build,
+ the compilation breaks. Reported by: snuffy Patch by: snuffy
+ (Closes issue #11313)
+
+2007-12-14 15:18 +0000 [r92938] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4
+ lines Up the length of the format on the SIP channel since it can
+ now be rather long. (closes issue #11552) Reported by:
+ francesco_r ........
+
+2007-12-14 15:14 +0000 [r92936] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 92933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007)
+ | 5 lines Change help documentation to match actual behavior
+ (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
+ (Closes issue #11548) ........
+
+2007-12-14 15:08 +0000 [r92935] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14
+ Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state
+ setting and overlap_task thread starting. ........
+
+2007-12-14 14:48 +0000 [r92913] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c,
+ res/res_features.c, apps/app_minivm.c, apps/app_amd.c,
+ res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+ main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert
+ ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy
+ (Closes issue #11547)
+
+2007-12-14 01:25 +0000 [r92876] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13
+ Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't
+ spam the CLI with messages about possible deadlocks. Instead just
+ print the intended single message every five seconds. (closes
+ issue 11537, reported and patched by dimas) ........
+
+2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: When working with dates, use numeric form
+ whenever possible, as it's faster. Also, a bunch of coding
+ guidelines fixes.
+
+ * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007)
+ | 5 lines Properly initialize polarity statuses, so that they are
+ detected properly. Reported by: julianjm Patch by: julianjm
+ (Closes issue #10238) ........
+
+2007-12-13 20:23 +0000 [r92811] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c,
+ apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the
+ new ast_module_user_* functions in a few documentation places.
+ (closes issue #11533) Reported by: IgorG Patches:
+ oldmacroclean.v1.diff uploaded by IgorG (license 20)
+
+2007-12-13 20:14 +0000 [r92810] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c, /: Merged revisions 92809 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1
+ line Make application help text a little more clear about the use
+ of extensions in a filename. ........
+
+2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec
+ 2007) | 3 lines Prevent another potential fd leak ........
+
+ * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec
+ 2007) | 3 lines Prevent a possible fd leak. ........
+
+2007-12-13 17:46 +0000 [r92779] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape
+ character, unless it really is an escape character.
+
+2007-12-13 16:23 +0000 [r92758] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Remove remnants of a poorly merged commit.
+ (92697)
+
+2007-12-13 15:40 +0000 [r92737] Doug Bailey <dbailey@digium.com>
+
+ * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to
+ local time zone
+
+2007-12-13 00:18 +0000 [r92697] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c:
+ Merged revisions 92696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600
+ (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file,
+ we previous continued on with what was already loaded. We do not
+ want to do this (see bug below for details). This makes it so
+ that if a [ is found without a ], the entire config will fail,
+ and nothing in it will be loaded. Issue 10690. ........
+
+2007-12-12 23:44 +0000 [r92676] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Revert an "optimization" that I added in
+ revision 89887, as the user who reported issue #11449 has
+ demonstrated that it actually was a performance hit on his
+ machine. I think that it is possible that it could still be a
+ benefit on systems under higher load, especially SMP systems, but
+ I don't have enough time or interest to find out at the moment.
+ (closes issue #11449)
+
+2007-12-12 21:22 +0000 [r92618] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600
+ (Wed, 12 Dec 2007) | 4 lines Don't increment user count until
+ after name has been recorded (if enabled). Issue 11048, tested by
+ pep. ........
+
+2007-12-12 20:05 +0000 [r92594] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, main/logger.c, main/utils.c,
+ apps/app_mixmonitor.c: Conversions of free to ast_free, where
+ applicable, and several other formatting fixes. Reported by:
+ eliel Patch by: eliel,tilghman (Closes issue #11209)
+
+2007-12-12 19:50 +0000 [r92562] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Merged revisions 92556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) |
+ 1 line resolve compiler warning ........
+
+2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_features.c: Same change to trunk as revision 92510. I'm
+ not sure why I merged this way, but I did.
+
+ * /: Blocked revisions 92510 via svnmerge ........ r92510 |
+ mmichelson | 2007-12-12 11:46:14 -0600 (Wed, 12 Dec 2007) | 7
+ lines Correctly detect where a dynamic feature was activated.
+ Before this patch, the channel which initiated the bridge was
+ always assumed to have been the one which activated the dynamic
+ feature. This patch corrects this. (closes issue #11529, reported
+ and patched by nic_bellamy) ........
+
+2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Correctly handle possible memory allocation
+ failure Reported by: eliel Patch by: eliel (Closes issue #11512)
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 92463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007)
+ | 4 lines Test directly for the API that fixed AST-2007-026, to
+ ensure that older versions of PostgreSQL are no longer
+ acceptable. (Closes issue #11526) ........
+
+2007-12-12 16:11 +0000 [r92444] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec
+ 2007) | 3 lines Removing an unused variable. ........
+
+2007-12-11 22:20 +0000 [r92423] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c,
+ channels/ringtone.h, include/asterisk/ulaw.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ include/asterisk/transcap.h, channels/misdn/isdn_lib.c,
+ channels/gentone.c, include/asterisk/zapata.h,
+ channels/misdn/isdn_lib.h, include/asterisk/doxyref.h,
+ channels/DialTone.h, channels/misdn/ie.c,
+ channels/misdn/chan_misdn_config.h, channels/iax2.h,
+ channels/misdn/portinfo.c, include/asterisk/udptl.h,
+ main/cygload.c, include/asterisk/translate.h: Doxygen updates,
+ formatting. misdn stuff needs a lot of doxygenification (Hello,
+ Qwell :-) )
+
+2007-12-11 22:10 +0000 [r92422] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Trunk build would fail due to the nonexistence of
+ zaptel hwgain structures missing. Patched configure to check for
+ this stuff and put a #ifdef around the offending code in
+ chan_zap. Thanks to file for overseeing this.
+
+2007-12-11 21:58 +0000 [r92421] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: We need to set the address we want to match
+ against before we actually do the match.. Closes issue #11518.
+
+2007-12-11 21:46 +0000 [r92402] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c: Removing a pointless memset. The memory
+ was just calloc'd, so the memory is already zeroed out
+
+2007-12-11 21:17 +0000 [r92401] Jason Parker <jparker@digium.com>
+
+ * apps/app_controlplayback.c: Add variable to show which key was
+ pressed to stop playback. Issue #11377, initial patch by johan.
+
+2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp <jcolp@digium.com>
+
+ * res/res_monitor.c: Only look to see if options are set if some
+ have been provided. (closes issue #11505) Reported by: Mike
+ Anikienko
+
+ * main/global_datastores.c, /: Merged revisions 92363 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec
+ 2007) | 6 lines Fix potential memory leak with the dialed
+ interfaces list if another memory allocation fails. (closes issue
+ #11507) Reported by: eliel Patches: global_datastores.c.patch
+ uploaded by eliel (license 64) ........
+
+2007-12-11 17:44 +0000 [r92324] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec
+ 2007) | 10 lines Fixing autofill to be more accurate.
+ Specifically, if calls ahead of the current caller were ringing
+ members (but not yet bridged) there could be available members
+ and waiting callers who would not get matched up. The member
+ availability checker was correctly determining the number of
+ available members in this scenario, but the queue itself did not
+ parallelly reflect this status on the pending calls. This commit
+ corrects the issue. (closes issue #11459, reported by
+ equissoftware, patched by me) ........
+
+2007-12-11 16:29 +0000 [r92305] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/unaligned.h, main/event.c: * In unaligned.h,
+ remove some unnecessary casts and mark the arg of the
+ get_unaligned functions as const * In event.c, use
+ get_unaligned_uint32() in a couple of places to fix issues on
+ architectures that don't allow unaligned access
+
+2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/devicestate.h, include/asterisk/agi.h,
+ include/asterisk/astobj2.h, include/asterisk/extconf.h,
+ include/asterisk/io.h, include/asterisk/cdr.h,
+ include/asterisk/aes.h, include/asterisk/_private.h,
+ include/asterisk/localtime.h, include/asterisk/hashtab.h,
+ include/asterisk/callerid.h, include/asterisk/logger.h,
+ include/asterisk/doxyref.h, include/asterisk/app.h,
+ include/asterisk/adsi.h, include/asterisk/event.h,
+ include/asterisk/causes.h, include/asterisk/alaw.h,
+ include/asterisk/ast_expr.h, include/asterisk/dsp.h,
+ include/asterisk/mod_format.h, include/asterisk/ael_structs.h,
+ include/asterisk/astdb.h: A lot of doxygen updates
+
+ * include/asterisk/frame.h: Doxygen updates
+
+2007-12-10 20:18 +0000 [r92243] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_zap.c: Add CLI commands to dynamically set hw and
+ sw gains
+
+2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp <jcolp@digium.com>
+
+ * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to
+ check_expr for platforms that need it. (closes issue #11484)
+ Reported by: snuffy
+
+ * /, main/rtp.c: Merged revisions 92204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6
+ lines Add G729A as another possible payload name for G729. Some
+ devices use this instead of G729, which is perfectly normal since
+ the payload number itself is defined and can't be used by
+ anything else so the name doesn't matter that much. (closes issue
+ #11483) Reported by: revolution Patches: rtp.diff uploaded by
+ revolution (license 346) ........
+
+2007-12-10 16:30 +0000 [r92203] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec
+ 2007) | 7 lines If there are no members in a queue, then the loop
+ where the datastore for detecting duplicate dialed numbers will
+ be skipped, meaning the datastore isn't created. This means that
+ when we try to free it, there's a crash. This stops that crash
+ from occurring. (closes issue #11499, reported by slavon, patched
+ by eliel) ........
+
+2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 92200 via svnmerge ........ r92200 | file |
+ 2007-12-10 12:13:43 -0400 (Mon, 10 Dec 2007) | 4 lines It is
+ possible for nativeformats to contain more then one codec, so
+ print out multiple ones. (closes issue #11366) Reported by: ovi
+ ........
+
+ * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1,
+ otherwise all PIDs greater than -1 will get the SIGHUP... and
+ that is bad. (closes issue #11453) Reported by: alanmcmillan
+
+2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Removing some LOG_DEBUG items
+
+ * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16
+ lines Avoid reinvite race situations with two Asterisks trying to
+ reinvite each other in 1.4 and trunk. This patch implements
+ support for the 491 error code that Asterisk 1.4 generates on
+ situations where we get an incoming INVITE and already has one in
+ progress. Thanks to mavetju for reporting and to Raj Jain for an
+ excellent explanation of the problem. Patch by myself. Tested
+ with 8 Asterisk servers connected to each other in a training
+ network. Closes issue #10481 ........
+
+ * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra
+ headers in the voicemail users listing in manager 1.1. Update
+ documentation too. (closes issue #11495) Reported by: caio1982
+ Patches: extra_vm_manager_info1.diff uploaded by caio1982
+ (license 22)
+
+2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo <rizzo@icir.org>
+
+ * build_tools/make_version, build_tools/make_version_h:
+ simplify/cleanup the scripts
+
+ * utils/Makefile: remove relative paths and use ASTTOPDIR instead.
+ Give a default value to ASTTOPDIR if unset so we can at least do
+ a 'make clean' without too much trouble. The proper fix, however,
+ is to partition the top level Makefile in a 'setup' and a 'main'
+ part, in a way that the 'setup' part can be included from
+ subdirs' Makefiles and allow targets to be built without going
+ through the top level Makefile.
+
+ * utils/clicompat.c: simplify this file
+
+ * doc/CODING-GUIDELINES: add a bit of info on the build
+ infrastructure
+
+ * Makefile: Fix the detection of modules installed from this build.
+ You can now add the path of local module subdirs from the command
+ line with make LOCAL_MOD_SUBDIRS= ....
+
+ * codecs/Makefile, apps/Makefile, Makefile.moddir_rules,
+ cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile,
+ formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules
+ the common instructions used to generate loadable and embedded
+ module lists. Individual Makefiles now are a lot simpler,
+ possibly as simple as this: -include
+ $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
+ MODULE_PREFIX=cdr_ all: _all include
+ $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because
+ in a single directory we can combine various types of modules
+ (app_, cdr_, func_, ... ) by simply listing them in the
+ MODULE_PREFIX variable. The individual Makefiles can also create
+ list of modules to be excluded by listing them in the variablel
+ MODULE_EXCLUDE (see an example in channels/Makefile). With this
+ change it becomes trivial to integrate a directory with locally
+ created/modified sources into the main build.
+
+ * Makefile, Makefile.moddir_rules: make the install target a bit
+ less noisy
+
+ * Makefile: document usage of several exported variables
+
+ * utils/Makefile: add hashtab.c to the list of files deleted
+
+ * Makefile.moddir_rules: another place where ../ should have been
+ ASTTOPDIR
+
+ * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile,
+ pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile,
+ funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR
+ and not .. to reference the top level directory.
+
+ * Makefile: Implement the outcome of a discussion on the -dev list
+ re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir
+ Cohen and Simon Perreault for extremely useful feedback: 1.
+ comment out the [directories] section the created asterisk.conf ;
+ you can set the correct defaults at build time using
+ INSTALL_PATH, so the repetition here is redundant and often
+ wrong. (The next step now is move asterisk.conf outside the
+ Makefile to asterisk.conf.sample, because there is little if
+ anything here that needs to be constructed at build/install
+ time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to
+ specify a path once if the two coincide. This should have no ill
+ side effects, because if you don't specify DESTDIR, you really
+ need INSTALL_PATH="" to set the correct defaults, and if you
+ specify DESTDIR the value is not overridden. The second part
+ required moving the 'export DESTDIR' right after the assignment
+ to prevent DESTDIR getting set by the export (this is documented
+ in the Makefile).o hopefully avoid the mistake)$ With this change
+ you can now do something like this from your source tree: make
+ INSTALL_PATH=/some/place install samples and then main/asterisk
+ -vdc which will pick up the correct config files and libraries
+ from /some/place - i.e. great for developers!
+
+ * main/config.c: remove unused code, and simplify the logic for
+ #include/#exec (still a lot of cleanup needed here).
+
+ * main/config.c: Implement comment_buffer and lline_buffer in terms
+ of the ast_str_*() API. I don't know if comment_buffers etc are
+ actually used at all...
+
+ * main/config.c: unify some common code
+
+ * main/config.c: normalize formatting
+
+ * main/config.c: document a nice technique to exit from a block in
+ case of errors.
+
+ * main/config.c: a little bit of documentation on how lines are
+ parsed.
+
+ * utils/ael_main.c: normalize header order, and add a comment on
+ the need to clean up this file.
+
+ * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need
+ netinet/in.h to be included before arpa/inet.h
+
+2007-12-07 23:32 +0000 [r91832-91891] Jason Parker <jparker@digium.com>
+
+ * /, main/dsp.c: Merged revisions 91890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600
+ (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the
+ input frame if we return a different frame in ast_dsp_process.
+ Issue 11273, pointed out by dimas, with a patch by eliel.
+ ........
+
+ * pbx/pbx_lua.c, configs/extensions.lua.sample: Update
+ documentation for pbx_lua. Closes issue #11492, patch by
+ mnicholson.
+
+2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 91830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) |
+ 5 lines Make the lock protecting each thread's list of locks it
+ currently holds recursive. I think that this will fix the
+ situation where some people have said that "core show locks"
+ locks up the CLI. (related to issue #11080) ........
+
+ * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07
+ Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code.
+ The ast_mutex_init() function had the mutex attribute object
+ marked as static. This means that multiple threads initializing
+ locks at the same time could step on each other and end up with
+ improperly initialized locks. (found when tracking down locking
+ issues related to issue #11080) ........
+
+ * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07
+ Dec 2007) | 6 lines I love fixing lock related errors in the lock
+ debugging code. That's about as ironic as it gets in Asterisk
+ programming land. Anyway, I spotted this bug while trying to
+ track down why systems are locking up and acting weird in issue
+ #11080. The mutex attribute object was marked as static in this
+ function when it should not have been. ........
+
+ * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) |
+ 6 lines * Add channel locking around datastore operations that
+ expect the channel to be locked. * Document why we don't record
+ Local channels in the dialed interfaces list. * Remove the dialed
+ variable as it isn't needed. * Restructure some code for clarity
+ and coding guidelines stuff ........
+
+2007-12-07 16:37 +0000 [r91782] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix a small typo in a comment. Closes issue
+ #11490
+
+2007-12-07 16:28 +0000 [r91781] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) |
+ 7 lines * Add channel locking around datastore operations that
+ expect the channel to be locked. * Document why we don't record
+ Local channels in the dialed interfaces list. * Handle memory
+ allocation failure. * Remove the dialed variable, as it wasn't
+ actually needed. * Tweak some formatting to conform to coding
+ guidelines. ........
+
+2007-12-07 16:11 +0000 [r91779] Jason Parker <jparker@digium.com>
+
+ * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c,
+ include/asterisk/pbx.h, main/cli.c: Add count of total number of
+ calls processed by asterisk during it's lifetime. Add number of
+ total calls and current calls to SNMP. Closes issue #10057, patch
+ by jcmoore.
+
+2007-12-07 16:11 +0000 [r91778] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 91777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) |
+ 6 lines * Add a bit more of a verbose comment as to why a hangup
+ frame needs to be queued up if autoservice gets a NULL return
+ from ast_read(). * Make the process of queueing the hangup frame
+ more efficient by putting the frame where it is going to end up
+ and avoiding some locking and extra memory allocations and
+ freeing. ........
+
+2007-12-07 15:40 +0000 [r91738] Mark Michelson <mmichelson@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 91737 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec
+ 2007) | 7 lines Hangups that happen during autoservice were not
+ processed appropriately. This is because a hangup actually causes
+ a NULL frame to be received, not a hangup frame. Queueing a
+ hangup if we receive a NULL frame during autoservice corrects
+ this problem (closes issue #11467, reported by jmls, patched by
+ me) ........
+
+2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) |
+ 2 lines Don't unlock the dialed_interfaces list until we're done
+ messing with the iterator. ........
+
+ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) |
+ 4 lines Allow dialing local channels from Queue() and Dial()
+ again. There was a slight flaw in the code to prevent call
+ forwards from looping that caused this problem. (related to issue
+ #11486) ........
+
+ * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) |
+ 7 lines Fix in an issue in the call forwarding handling code that
+ was causing crashes on every call into a queue. I'm not entirely
+ sure about the logic in this part of the code, so I want to look
+ at it some more tomorrow. However, this makes it safe and keeps
+ it from crashing. (closes issue #11486, reported by adamg,
+ patched by me) ........
+
+2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/rtp.c: Merged revisions 91637 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007)
+ | 5 lines At the end of a call, when we're reporting, RTCP may
+ already be partially torn down, so check for NULL dereference
+ Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
+ ........
+
+ * channels/chan_zap.c: Add a manager event for PRI events: this
+ will help manager users detect when a D-channel goes down
+
+ * main/cdr.c: If duration or billsec are not yet calculated,
+ calculate them on demand.
+
+2007-12-06 21:57 +0000 [r91598] Jason Parker <jparker@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3
+ cdr module.. Closes issue #11070, patch by seanbright.
+
+2007-12-06 21:03 +0000 [r91579] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Handle allocation failure of the heard and
+ deleted arrays of the vm_state. (closes issue #11408, reported
+ and patched by jaroth)
+
+2007-12-06 20:52 +0000 [r91561] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007)
+ | 3 lines Properly escape cdr->src and cdr->dst and ensure we use
+ thread-safe escaping (Fixes AST-2007-026) ........ r90736 |
+ tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines
+ If both dbhost and dbsock were not set, a NULL deref could result
+ Reported by: xrg Patch by: tilghman (Closes issue #11387)
+ ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
+ Dec 2007) | 5 lines Solaris requires the inclusion of
+ sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
+ snuffy,tilghman (Closes issue #11430) ........
+
+2007-12-06 19:12 +0000 [r91502] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 91501 via svnmerge ........ r91501 | russell
+ | 2007-12-06 13:11:35 -0600 (Thu, 06 Dec 2007) | 5 lines Add a
+ new module flag to indicate that a build sum is present. Modules
+ built against older Asterisk 1.4 headers will now load properly
+ with just a warning indicating that they are old and may cause
+ problems. (patch by paravoid) ........
+
+2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Make sure we clear these flags when libpri
+ is not installed
+
+2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp <jcolp@digium.com>
+
+ * main/udptl.c, /: Merged revisions 91450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6
+ lines Fix various in the udptl implementation. It could return
+ empty modem frames, have an incorrect sequence number on packets,
+ and display the wrong sequence number in the debug messages.
+ (closes issue #11228) Reported by: Cache Patches: udptl-4.patch
+ uploaded by dimas (license 88) ........
+
+ * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4
+ lines Add support for accepting and sending T.38 in the initial
+ INVITE. (closes issue #9402) Reported by: thdei ........
+
+2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson <oej@edvina.net>
+
+ * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation
+ for the massive manager changes to manager version 1.1 -
+ hopefully a more consistent manager interface.
+
+ * main/manager.c: - The Ping Action - Now use Response: success -
+ New header "Ping: pong" :-) - The Events action - Now use
+ Response: Success - The new status is reported as "Events: On" or
+ "Events: Off" - Report if manager is enabled in the reload event
+ Small cleanups... From moremanager
+
+ * main/channel.c: Changes to manager events in channel.c - Newstate
+ event - Now has "CalleridNum" for numeric caller id, like
+ Newchannel - The event does not send "<unknown>" for unknown
+ caller IDs just an empty field - Newstate and Newchannel events -
+ these have changed headers "State" -> ChannelStateDesc Text based
+ channel state -> ChannelState Numeric channel state - The events
+ does not send "<unknown>" for unknown caller IDs just an empty
+ field - Newstate event - Now has "CalleridNum" for numeric caller
+ id, like Newchannel - The event does not send "<unknown>" for
+ unknown caller IDs just an empty field - Link and Unlink events -
+ The "Link" and "Unlink" bridge events in channel.c are now
+ renamed to "Bridge" - The link state is in the bridgestate:
+ header as "Link" or "Unlink" - For channel.c bridges,
+ "Bridgetype: core" is added. This opens up for bridge events in
+ rtp.c and channel drivers - The "Rename" manager event has a
+ renamed header, to use the same terminology for the current
+ channel as other events - Oldname -> Channel (Moremanager)
+
+ * main/cdr.c: New manager event when a channel changes account
+ code. Maybe belongs in the new cdr category? ---moremanager---
+ Event: NewAccountCode Modules: cdr.c Purpose: To report a change
+ in account code for a live channel Example: Event: NewAccountCode
+ Privilege: call,all Channel: SIP/olle-01844600 Uniqueid:
+ 1177530895.2 AccountCode: Stinas account 1234848484
+ OldAccountCode: Olles Account 12345
+
+ * apps/app_dial.c: - Dial event - Event Dial has new headers, to
+ comply with other events - Source -> Channel Channel name
+ (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring
+ Dialstring in app data (moremanager)
+
+ * apps/app_meetme.c: Adding small missing but important comma...
+
+ * apps/app_meetme.c: A big oops...
+
+ * apps/app_meetme.c: The MeetmeJoin now has caller ID name and
+ Caller ID number fields (like MeetMeLeave) (Moremanager)
+
+ * channels/chan_zap.c: Update ZapShowChannels so that you can
+ specify one channel. Action ZapShowChannels Header changes -
+ Channel: -> ZapChannel For active channels, the Channel: and
+ Uniqueid: headers are added You can now add a "ZapChannel: "
+ argument to zapshowchannels actions to only get information about
+ one channel. From the moremanager branch
+
+ * main/logger.c: Doxygen updates
+
+ * include/asterisk/logger.h, /, main/logger.c, main/loader.c:
+ Merged revisions 91366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4
+ lines Make sure logger is reloaded at general reload in the cli.
+ (Discovered during Asterisk training in Portugal) ........
+
+ * main/manager.c: Change description of new manager command
+
+ * main/manager.c, CHANGES: Add manager command for showing all
+ current channels. Thanks, eliel, for writing the original patch.
+ Modified by me to follow other manager events and the new
+ "moremanager" style. (closes issue #11478) Reported by: eliel
+ Patches: manager.c.patch uploaded by eliel (license 64)
+
+2007-12-06 04:37 +0000 [r91328] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Instead of iterating through the entire epoll
+ events array just look at the ones that will actually contain
+ data. (props to eliel on IRC for this)
+
+2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec
+ 2007) | 3 lines Reverting extra stuff I didn't mean to commit
+ ........
+
+ * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec
+ 2007) | 11 lines The 'G' option for Dial() did not properly
+ handle the case where only a label was provided. This was due to
+ the fact that the answering channel did not have an extension
+ set, so ast_parseable_goto would fail. This fix eliminates the
+ call to ast_parseable_goto on the answering channel since it is a
+ wasteful call. The answering channel and the calling channel are
+ both directed to the same extension and context, just different
+ priorities, so we can just copy the values from the calling
+ channel to the answering channel and increment the answering
+ channel's priority. (closes issue #11382, reported by jon, patch
+ by me with correction by jon) ........
+
+2007-12-05 21:46 +0000 [r91238] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 91237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007)
+ | 2 lines Upgrade to the latest version of extra sounds ........
+
+2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant <russell@digium.com>
+
+ * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) |
+ 10 lines Make the lock in the threadstorage debugging code
+ untracked to avoid a deadlock on thread destruction. (closes
+ issue #11207) Reported by: ys Patches: threadstorage.c.diff
+ uploaded by ys (license 281) Also fixes an open bug report:
+ (closes issue #11446) ........
+
+ * apps/app_directory.c: Resolve compiler warnings.
+
+2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, UPGRADE.txt, configs/manager.conf.sample,
+ CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change
+ cdr_manager to use a "CDR" level, rather than the (overcrowded)
+ "call" level. (Closes issue #11015)
+
+ * CHANGES, apps/app_directory.c: Added multiple name listing.
+ (Closes issue #10413)
+
+2007-12-05 16:14 +0000 [r91171] Joshua Colp <jcolp@digium.com>
+
+ * configs/http.conf.sample: Remove second prefix line. Only need it
+ documented once in the same file. (closes issue #11472) Reported
+ by: eserra Patches: http.conf.sample.diff uploaded by eserra
+ (license 45)
+
+2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename
+ "username" to "defaultuser" to match with "defaultip". "Username"
+ still works, but is deprecated.
+
+ * channels/chan_sip.c: Remove the cseqs from "sip show channel" and
+ make more place for the call ID.
+
+2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: revert part of my changes from earlier today
+ since this code is no longer dependent on libpri.h
+
+2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant <russell@digium.com>
+
+ * res/res_odbc.c: Use ast_free() instead of free(). (closes issue
+ #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded
+ by Laureano (license 265)
+
+ * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04
+ Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were
+ reported as happening on Mac systems. It turns out that the
+ problem was the Mac version of the ast_atomic_fetchadd_int()
+ function. The Mac atomic add function returns the _new_ value,
+ while this function is supposed to return the old value. So, the
+ crashes happened on unreferencing objects. If the reference count
+ was decreased to 1, ao2_ref() thought that it had been decreased
+ to zero, and called the destructor. However, there was still an
+ outstanding reference around. (closes issue #11176) (closes issue
+ #11289) ........
+
+ * /, main/utils.c: Merged revisions 91074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) |
+ 4 lines When DEBUG_THREADS is enabled, we only have the details
+ about who is holding a lock that we are waiting on for a mutex,
+ not rwlocks. This should fix the problem where people have
+ reported "core show locks" crashing sometimes. ........
+
+ * channels/chan_zap.c: Fix mwimonitornotify on reload ... again.
+ This option was only read at startup so a reload would erase it
+ and not reset it. (pointed out by tzafrir)
+
+ * /: Blocked revisions 91032 via svnmerge ........ r91032 | russell
+ | 2007-12-04 17:46:40 -0600 (Tue, 04 Dec 2007) | 5 lines Modify
+ file.h to maintain API compatibility with earlier versions. If a
+ recent compiler is being used, then a warning will show up for
+ any modules still using the old name "private" instead of
+ "_private". (patch suggested by paravoid) ........
+
+ * utils/astman.c: Fix the build of astman. Any file that includes
+ any asterisk sub-headers needs to first include asterisk.h.
+ (closes issue #11394)
+
+2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Don't error when we don't have libpri
+ installed with libss7 support. Also, print the debug message
+ anyway if we can't find the right PRI
+
+2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 90967 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) |
+ 7 lines Make some changes to some additions I made recently for
+ doing channel autoservice when looking up extensions. This code
+ was added to handle the case where a dialplan switch was in use
+ that could block for a long time. However, the way that I added
+ it, it did this for all extension lookups. However, lookups in
+ the in-memory tree of extensions should _not_ take long enough to
+ matter. So, move the autoservice stuff to be only around
+ executing a switch. ........
+
+ * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I
+ guess I only added this line in my head. (thanks to tzafrir for
+ pointing it out)
+
+2007-12-04 21:46 +0000 [r90993] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue
+ #11412)
+
+2007-12-04 21:23 +0000 [r90991] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Add manager action
+ 'sipshowregistry'. Closes issue #11464, patch by eliel.
+
+2007-12-04 19:08 +0000 [r90949] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/callerid.h, channels/chan_zap.c,
+ main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support
+ for monitoring MWI on FXO lines. This introduces two new options
+ for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor
+ option enables MWI monitoring. When the MWI state on a line
+ changes, then the script specified by mwimonitornotify will be
+ executed for custom handling of the state change, similar to the
+ externnotify option of voicemail.conf. Also, when the MWI state
+ on an FXO line changes, an internal Asterisk event is generated
+ to indicate the new state of the associated mailbox. That may,
+ any module that cares about MWI information will get notified and
+ can handle it just as if app_voicemail had sent this
+ notification. (BE-253, original patch from markster, with some
+ minor modifications by me to add comments, documentation, and
+ internal event support)
+
+2007-12-04 18:29 +0000 [r90930] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Kevin suggested doing the reverse of my
+ last commit, since imap_retrieve_file does not modify the
+ contents of the "mailbox" string. In other words, I'm changing
+ the imap_retrieve_file function to take a const char* as the
+ third argument so that I don't need to cast const char*'s as
+ char*'s to suppress compiler warnings.
+
+2007-12-04 18:15 +0000 [r90929] Jason Parker <jparker@digium.com>
+
+ * Makefile: Add Makefile alias target 'pdf' which does the same
+ thing as asterisk.pdf. Issue 11452, reported by blitzrage.
+
+2007-12-04 18:14 +0000 [r90928] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Suppress a compiler warning due to
+ discarding a "const" qualifier
+
+2007-12-04 18:09 +0000 [r90927] Jason Parker <jparker@digium.com>
+
+ * main/global_datastores.c: Fix build, that some people aren't
+ seeing for some reason.
+
+2007-12-04 17:51 +0000 [r90899] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Wrong locking style got merged from 1.4 to
+ trunk. My mistake.
+
+2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: fix build of this module when libpri and/or
+ libss7 are or are not present
+
+2007-12-04 17:38 +0000 [r90879] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 90876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600
+ (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after
+ allocating a timing fd, we need to make sure to close it. Issue
+ 11454, patch by eliel. ........
+
+2007-12-04 17:36 +0000 [r90878] Russell Bryant <russell@digium.com>
+
+ * main/Makefile: Fix a silly little typo :)
+
+2007-12-04 17:35 +0000 [r90877] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but
+ blocked in trunk since this hadn't been merged yet.
+
+2007-12-04 17:08 +0000 [r90873] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, main/global_datastores.c (added),
+ channels/chan_local.c, /, main/Makefile,
+ include/asterisk/channel.h, include/asterisk/global_datastores.h
+ (added), apps/app_queue.c: Merged revisions 90735 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03
+ Dec 2007) | 22 lines A big one... This is the merge of the
+ forward-loop branch. The main change here is that call-forwards
+ can no longer loop. This is accomplished by creating a datastore
+ on the calling channel which has a linked list of all devices
+ dialed. If a forward happens, then the local channel which is
+ created inherits the datastore. If, through this progression of
+ forwards and datastore inheritance, a device is attempted to be
+ dialed a second time, it will simply be skipped and a warning
+ message will be printed to the CLI. After the dialing has been
+ completed, the datastore is detached from the channel and
+ destroyed. This change also introduces some side effects to the
+ code which I shall enumerate here: 1. Datastore inheritance has
+ been backported from trunk into 1.4 2. A large chunk of code has
+ been removed from app_dial. This chunk is the section of code
+ which handles the call forward case after the channel has been
+ requested but before it has been called. This was removed because
+ call-forwarding still works fine without it, it makes the code
+ less error-prone should it need changing, and it made this set of
+ changes much less painful to just have the forwarding handled in
+ one place in each module. 3. Two new files, global_datastores.h
+ and .c have been added. These are necessary since the datastore
+ which is attached to the channel may be created and attached in
+ either app_dial or app_queue, so they need a common place to find
+ the datastore info. This approach was taken in case similar
+ datastores are needed in the future, there will be a common place
+ to add them. ........
+
+2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson <oej@edvina.net>
+
+ * apps/app_queue.c: (closes issue #11431) Reported by: Laureano
+ Patches: app_queue.c.patch uploaded by Laureano (license 265)
+
+ * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel
+ Patches: core.show.hint.patch uploaded by eliel (license 64)
+
+ * CHANGES: (closes issue #11462) Reported by: eliel Patches:
+ CHANGES.patch uploaded by eliel (license 64)
+
+2007-12-04 15:01 +0000 [r90851] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Pass the Asterisk version to AGI scripts as part
+ of the initial dump of info Reported by: acunningham Patch by:
+ acunningham (Closes issue #11398)
+
+2007-12-04 11:50 +0000 [r90834] Luigi Rizzo <rizzo@icir.org>
+
+ * res/Makefile: fix build on cygwin
+
+2007-12-04 05:31 +0000 [r90799] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 90798 via svnmerge ........ r90798 | file |
+ 2007-12-04 01:29:33 -0400 (Tue, 04 Dec 2007) | 2 lines Fix build
+ issue on the build cluster. ........
+
+2007-12-03 23:52 +0000 [r90760] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
+ Dec 2007) | 5 lines Solaris requires the inclusion of
+ sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
+ snuffy,tilghman (Closes issue #11430) ........
+
+2007-12-03 23:49 +0000 [r90746] Steve Murphy <murf@digium.com>
+
+ * main/hashtab.c: A small fix from snuffy
+
+2007-12-03 23:48 +0000 [r90738] Jason Parker <jparker@digium.com>
+
+ * res/res_monitor.c: Add manager events for when a monitor is
+ started or stopped. Closes issue #10191, patch by dgradecak.
+
+2007-12-03 23:29 +0000 [r90737] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03
+ Dec 2007) | 5 lines If both dbhost and dbsock were not set, a
+ NULL deref could result Reported by: xrg Patch by: tilghman
+ (Closes issue #11387) ........
+
+2007-12-03 22:07 +0000 [r90697] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600
+ (Mon, 03 Dec 2007) | 4 lines Make sure we always close the
+ conference fd if we have an open one. Issue 11383, reported by
+ markmhy, patch by eliel. ........
+
+2007-12-03 21:24 +0000 [r90670] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Replacing some calls to free() with
+ ast_free(). (closes issue #11448, reported and patched by jaroth)
+
+2007-12-03 21:03 +0000 [r90656] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands
+ for speech recognition. These mirror the dialplan applications
+ mostly but present the information in a nicer fashion. The SPEECH
+ RECOGNIZE command for example will return the results instead of
+ having to query the dialplan functions.
+
+2007-12-03 21:00 +0000 [r90644] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec
+ 2007) | 5 lines Changing some bad logic when calculating the
+ interdigit timeout. (closes issue #11402, reported and patched by
+ eferro) ........
+
+2007-12-03 20:58 +0000 [r90631] Jason Parker <jparker@digium.com>
+
+ * /, res/res_features.c: Merged revisions 90607 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600
+ (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce
+ application. Issue #11436, reported by lytledd, patch by eliel.
+ ........
+
+2007-12-03 20:30 +0000 [r90591] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Avoid an additional function call. Reported by:
+ eliel Patch by: eliel (Closes issue #11438)
+
+2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 90588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2
+ lines Do not create a smoother for G723.1 frames, they need to be
+ left alone to their native 20/24 byte size. ........
+
+ * main/channel.c, /, include/asterisk/channel.h, .cleancount:
+ Merged revisions 90548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2
+ lines Preserve the indication currently playing on a channel when
+ a masquerade operation happens. (issue #BE-88) ........
+
+2007-12-03 18:21 +0000 [r90547] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 90546 via svnmerge ........ r90546 | qwell |
+ 2007-12-03 12:20:49 -0600 (Mon, 03 Dec 2007) | 4 lines Only log
+ debug messages if debug is enabled. Closes issue #11416, patch by
+ casper. ........
+
+2007-12-03 16:46 +0000 [r90528] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample: Updating sample queues.conf file to
+ show how multiple periodic announcements may be specified since
+ this was not documented previously (closes issue #11432, reported
+ and patched by Laureano)
+
+2007-12-03 14:14 +0000 [r90508] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Remove the file descriptors from the main poll
+ channel when the channel is hung up during the dialing attempt,
+ and make sure a channel exists before trying to remove it at the
+ end. (closes issue #11441) Reported by: blitzrage
+
+2007-12-02 18:20 +0000 [r90471] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) |
+ 6 lines The other day when I went through making changes as a
+ result of the ao2_link() change, I added some code to set
+ pointers to NULL after they were unreferenced. This pointed out
+ that in this place, the object was unreferenced before the code
+ was done using it. So, move the unref down a little bit. (crash
+ reported by jmls on IRC) ........
+
+2007-12-02 09:42 +0000 [r90433] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 90432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007)
+ | 7 lines Clarify the return value on autoservice. Specifically,
+ if you started autoservice and autoservice was already on, it
+ would erroneously return an error. Reported by: adiemus Patch by:
+ dimas (Closes issue #11433) ........
+
+2007-12-01 01:37 +0000 [r90410] Jason Parker <jparker@digium.com>
+
+ * res/res_adsi.c: Only reload if the config file has changed.
+ Closes issue #11281, patch by eliel.
+
+2007-11-30 21:19 +0000 [r90388] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, include/asterisk/app.h,
+ include/asterisk/audiohook.h, res/res_features.c,
+ include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c,
+ configs/features.conf.sample: Adding support for the
+ "automixmonitor" dial and queue options. This works in much the
+ same way as the automonitor, except that instead of using the
+ monitor app, it uses the mixmonitor app. By providing an 'x' or
+ 'X' as a dial or queue option, a DTMF sequence may be entered (as
+ defined in features.conf) to start the one-touch mixmonitor. This
+ patch also introduces some new API calls to the audiohooks code
+ for searching for an audiohook by type and for searching for a
+ running audiohook by type. Big thanks to joetester for writing
+ the initial patch, testing it and patiently waiting for it to be
+ committed. (closes issue #10185, reported and patched by
+ xmarksthespot)
+
+2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged
+ revisions 90348 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) |
+ 8 lines Change the behavior of ao2_link(). Previously, in
+ inherited a reference. Now, it automatically increases the
+ reference count to reflect the reference that is now held by the
+ container. This was done to be more consistent with ao2_unlink(),
+ which automatically releases the reference held by the container.
+ It also makes it so it is no longer possible for a pointer to be
+ invalid after ao2_link() returns. ........
+
+ * /, include/asterisk/astobj2.h: Merged revisions 90310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) |
+ 2 lines Add some notes on the behavior of ao2_unlink() after a
+ discussion with Tilghman ........
+
+2007-11-30 14:45 +0000 [r90270] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6
+ lines Fix locking issues under one legged replaces scenarios.
+ (closes issue #11420) Reported by: irroot Patches:
+ chan_sip_oneleg.patch uploaded by irroot (license 52) ........
+
+2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov
+ 2007) | 5 lines Clear the DTMF buffer if the call times out.
+ (closes issue #11418, reported and patched by eferro) ........
+
+ * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov
+ 2007) | 6 lines This patch handles the case where a queue member
+ with a negative penalty is added via the manager. If a negative
+ value is submitted for a member penalty, we set it to 0. (closes
+ issue #11411, reported and patched by Laureano) ........
+
+2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29
+ Nov 2007) | 2 lines Properly escape input buffers (Fixes
+ AST-2007-025) ........
+
+ * /, formats/format_wav.c, formats/format_pcm.c,
+ formats/format_ogg_vorbis.c, main/file.c,
+ include/asterisk/mod_format.h, formats/format_h263.c,
+ formats/format_h264.c, formats/format_wav_gsm.c,
+ formats/format_g726.c: Merged revisions 90155 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007)
+ | 5 lines Use of "private" as a field name in a header file
+ messes with C++ projects Reported by: chewbacca Patch by: casper
+ (Closes issue #11401) ........
+
+ * include/asterisk/lock.h: Fix build of trunk
+
+ * /, sounds/Makefile: Merged revisions 90154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007)
+ | 2 lines Upgrade the core sounds release version ........
+
+2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/Makefile, utils/hashtest.c: let's try this again... *all*
+ compilation and linking in Asterisk should be done using the
+ standard compilation rules, not manually created ones. changing
+ hashtest.c to use these rules caused the compiler to notice a
+ large number of coding guidelines violations, so those are fixed
+ too.
+
+ * main/manager.c: restore behavior from the 1.4 branch... manager
+ users created via users.conf should default to *all* permissions,
+ not none
+
+2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 90147 via svnmerge ........ r90147 | russell
+ | 2007-11-28 18:36:59 -0600 (Wed, 28 Nov 2007) | 1 line fix some
+ formatting i accidentally changed ........
+
+ * main/channel.c, /, include/asterisk/channel.h,
+ funcs/func_callerid.c: Merged revisions 90145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) |
+ 5 lines This set of changes is to make some callerID handling
+ thread-safe. The ast_set_callerid() function needed to lock the
+ channel. Also, the handlers for the CALLERID() dialplan function
+ needed to lock the channel when reading or writing callerid
+ values directly on the channel structure. ........
+
+ * include/asterisk/file.h, /, main/file.c: Merged revisions 90142
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) |
+ 4 lines Merge a change from team/russell/chan_refcount ... This
+ makes ast_stopstream() thread-safe. ........
+
+ * include/asterisk/audiohook.h: Merge another small doxygen change
+ from team/russell/chan_refcount to indicate that a channel
+ doesn't need to be locked before calling a certain function.
+
+ * include/asterisk/channel.h: Merge some channel.h doxygen updates
+ from team/russell/chan_refcount This was mostly to note whether a
+ channel needed to be locked or not before calling these
+ functions. However, I added some other things, too.
+
+2007-11-28 23:03 +0000 [r90102] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions
+ 90101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6
+ lines Fix a few memory leaks. (closes issue #11405) Reported by:
+ eliel Patches: load_realtime.patch uploaded by eliel (license 64)
+ ........
+
+2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/users.conf.sample, main/manager.c, /: Merged revisions
+ 90098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007)
+ | 2 lines it is impossible to set permissions for manager
+ accounts created by users.conf (reported internally, patched by
+ me) ........
+
+2007-11-28 22:32 +0000 [r90099] Joshua Colp <jcolp@digium.com>
+
+ * main/cli.c: file says... compile before you commit!
+
+2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Removing a pointless check of option_debug
+
+ * main/pbx.c, /: Merged revisions 90059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov
+ 2007) | 13 lines Removing some seemingly pointless code. This
+ sets a channel variable for every priority executed in the
+ dialplan if you have debug set to anything non-zero. This seems
+ pointless due to the fact that these channel variables are not
+ referenced anywhere else in the code and their names are esoteric
+ enough that they would not be practical to reference in the
+ dialplan. Plus the fact that this behavior isn't documented
+ anywhere means that the change is not likely to cause any
+ disruption. If anything, this may actually cause a slight
+ performance increase if running with debug on. The motivating
+ influence for this code change is the eventwhencalled option for
+ queues. If set to vars, all channel variables will be output to
+ the manager. These unnecessary channel variables make the output
+ a lot more difficult to deal with. ........
+
+2007-11-28 20:33 +0000 [r90039] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit
+ transparent
+
+2007-11-28 20:27 +0000 [r90038] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c:
+ Remove "old"-style CLI handler, since nothing uses it anymore.
+ Closes issue #11403, patch by eliel. This also completes the
+ janitor project.
+
+2007-11-28 17:37 +0000 [r90000] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 89999 via svnmerge ........ r89999 |
+ mmichelson | 2007-11-28 11:30:47 -0600 (Wed, 28 Nov 2007) | 6
+ lines Recording greetings when using IMAP storage was causing
+ zero-length files to be stored. Since greetings are not retrieved
+ from IMAP anyway, it is pointless to attempt storing them there.
+ (closes issue #11359, reported by spditner, patched by me)
+ ........
+
+2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp <jcolp@digium.com>
+
+ * main/cli.c: Hide CLI commands starting with _ from tab completion
+ as was done previously. (closes issue #11395) Reported by: eliel
+ Patches: cli.c.patch uploaded by eliel (license 64)
+
+ * main/abstract_jb.c, res/res_agi.c: Fix a few log messages.
+ (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff
+ uploaded by IgorG (license 20)
+
+2007-11-28 00:49 +0000 [r89947] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: Merge some little changes from
+ team/russell/chan_refcount to help reduce the diff to trunk. This
+ just removes some checks on the return value of alloca(), as
+ behavior is undefined if it runs out of stack space, and we don't
+ check it anywhere else.
+
+2007-11-28 00:47 +0000 [r89946] Mark Michelson <mmichelson@digium.com>
+
+ * configs/musiconhold.conf.sample, configs/extconfig.conf.sample,
+ res/res_musiconhold.c, CHANGES: Adding support for realtime music
+ on hold. The following are the main points: 1. When moh is
+ started, we search first in memory to find the class. If we do
+ not find it in memory, we search realtime instead. 2. When moh is
+ restarted (as in, it had been started on this particular channel,
+ stopped, and now we're starting it again), if using the "files"
+ mode, then realtime will always be rechecked. If you are using
+ other modes, however, we will simply reattach to the external
+ running process which was playing moh earlier in the call. This
+ is a necessary compromise so that we don't end up with too many
+ background processes. 3. musiconhold.conf has a general section
+ now. It has one option: cachertclasses. If set to yes, then moh
+ classes found in realtime will be added to the in-memory list.
+ This has the advantage of not requiring database lookups each
+ time moh is started, but it has the disadvantage of not truly
+ being realtime. I have tested this for functionality, and it
+ passes. I also tested this under valgrind and there are no memory
+ problems reported under typical use. Special thanks to Sergee for
+ implementing this feature and enduring my complaints on the
+ bugtracker! (closes issue #11196, reported and patched by sergee)
+
+2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) |
+ 4 lines - update documentation for some of the goto functions to
+ note that they handle locking the channel as needed - update
+ ast_explicit_goto() to lock the channel as needed ........
+
+ * include/asterisk/channel.h: Document that the channel is not
+ locked when the send_digit_begin and end callbacks get called.
+
+ * main/autoservice.c, /: Merged revisions 89886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) |
+ 2 lines Don't do frame processing if ast_read() returned NULL.
+ ........
+
+ * channels/chan_iax2.c: Merge changes from
+ team/russell/iax2_frame_queue This patch is an optimization for
+ chan_iax2. This module is now heavily multi-threaded. However,
+ there is still a good number of globally shared resources that
+ prevent things from happen asynchronously. One of those things
+ was the global IAX frame queue. This queue was used to hold
+ frames that have been deferred for transmitting by another
+ thread, and frames that may need to get retransmitted. I changed
+ the frame queue to be per-call, since almost all of the frame
+ queue handling only cares about frames specific to a call number.
+
+ * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) |
+ 3 lines Instead of depending on the return value of ast_true(),
+ explicitly set the eventwhencalled variable to 1. ........
+
+ * main/pbx.c, /: Merged revisions 89839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) |
+ 2 lines Don't start/stop autoservice in pbx_extension_helper()
+ unless a channel exists ........
+
+2007-11-27 23:11 +0000 [r89838] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov
+ 2007) | 12 lines Two changes with regards to the
+ 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
+ unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
+ did exactly the same thing. Thus the sign change of the ast_true
+ call. 2) The vars2manager function overwrote a \n for every
+ channel variable it parsed, resulting in bizarre output for the
+ channel variables. This patch remedies this. (related to issue
+ #11385, however I'm not sure if this will actually be enough to
+ close it) ........
+
+2007-11-27 22:42 +0000 [r89835] Russell Bryant <russell@digium.com>
+
+ * channels/chan_misdn.c: Bring in a small change from
+ team/russell/chan_refcount This replaces tab completion code with
+ the use of a public function that does the same thing
+
+2007-11-27 22:14 +0000 [r89792] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the
+ conditional unlock of the contexts when the hash table is used
+ instead; also, used the ast_free_ptr as advised.
+
+2007-11-27 22:05 +0000 [r89791] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) |
+ 41 lines Merge changes from team/russell/autoservice_1.4 This set
+ of changes fixes an issue that was reported to me on IRC
+ yesterday. The user, d1mas, was using chan_zap for incoming calls
+ and was having DTMF recognition issues in some situations.
+ Specifically, he noticed that the problem occurred when using
+ DISA or WaitExten. He also noticed that when using Read, the
+ problem did not occur. His system also used DUNDi for dialplan
+ lookups. So, he theorized that if the DUNDi lookups blocked for
+ some period of time, that audio from the zap channel could get
+ lost. If the audio got lost, then it wouldn't be run through the
+ DTMF detector, and digits could get lost. He was correct, and the
+ following set of changes fixes the problem. However, the changes
+ go a little bit further than what was necessary to fix this exact
+ problem. 1) I updated pbx_extension_helper() to autoservice the
+ associated channel to handle cases where extension lookups may
+ take a long time. This would normally be a dialplan switch that
+ does some lookup over the network, such as the DUNDi or IAX2
+ switches. This ensures that even while a DUNDi lookup is
+ blocking, the channel will be continuously serviced. 2) I made a
+ change to the autoservice code. This is actually something that
+ has bothered me for a long time. When a channel is in
+ autoservice, _all_ frames get thrown away. However, some frames
+ really shouldn't be thrown away. The most notable examples are
+ signalling (CONTROL) frames, and DTMF. So, this patch queues up
+ important frames while a channel is in autoservice. When
+ autoservice is stopped on the channel, the queued up frames get
+ stuck back on the channel so that they can get processed instead
+ of thrown away. 3) I made another change to the autoservice code
+ to handle the case where autoservice is started on channels
+ recursively. Previously, you could call ast_autoservice_start()
+ multiple times on a channel, and it would stop the first time
+ ast_autoservice_stop() gets called. Now, it will ensure that
+ autoservice doesn't actually stop until the final call to
+ ast_autoservice_stop(). ........
+
+2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson <oej@edvina.net>
+
+ * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A
+ few more "moremanager" fixes
+
+ * include/asterisk.h, main/asterisk.c, main/loader.c: More
+ "moremanager" fixes. Manager commands to check module status.
+
+ * include/asterisk/manager.h: More "moremanager" changes - doxygen
+ docs and changing manager version (finally) before making more
+ dramatic changes.
+
+ * channels/chan_iax2.c: More additions from the "moremanager"
+ branch, this time for IAX2.
+
+2007-11-27 20:24 +0000 [r89733] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocked revisions 89727 via svnmerge ........ r89727 |
+ mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6
+ lines Changing some calls from free() to ast_free() since they
+ were allocated with ast_calloc(). (closes issue #11390, reported
+ and patched by Laureano) ........
+
+2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/app.c: Merged revisions 89709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007)
+ | 2 lines on second thought... revert all the other changes i've
+ made in app options parsing leaving only one: if an empty
+ argument is supplied for an option, set that argument pointer to
+ point to an empty string rather than NULL, so that the
+ application can do normal checks on it without worrying about it
+ being NULL ........
+
+2007-11-27 20:17 +0000 [r89710] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: remove a duplicate manager event
+
+2007-11-27 19:50 +0000 [r89706] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_gtalk.c: Manager events from the "moremanager"
+ branch
+
+2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/app.c: Merged revisions 89701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007)
+ | 2 lines generate a warning when an application option that
+ requires an argument is ignored due to lack of an argument
+ ........
+
+2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Starting to merge changes from the
+ "moremanager" branch. Documentation will follow.
+
+ * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The
+ following patch with updates for trunk. Works much better in
+ trunk. Also by accident fixed a bad typo by a previous committer,
+ which actually made video calls not work fully... Merged
+ revisions 89630 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12
+ lines If we get a codec offer using a well-known payload type,
+ but using it for another codec that we don't know, Asterisk did
+ not remove that codec from the list. With this patch, we remove
+ the codec from audio and video rtp objects and deny it ever
+ existed. Thanks to lasse for testing. (closes issue #11376)
+ Reported by: lasse Patches: bug11376.txt uploaded by oej (license
+ 306) Tested by: lasse ........
+
+2007-11-27 19:12 +0000 [r89683] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/strings.h: Add an S_COR macro, which is similar
+ to the existing S_OR macro, except with an additional boolean
+ arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes:
+ S_COR(foo, bar, "baz")
+
+2007-11-27 18:50 +0000 [r89682] Steve Murphy <murf@digium.com>
+
+ * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11,
+ pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14,
+ pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
+ pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
+ pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22,
+ res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex,
+ pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8:
+ made AEL 8-bit transparent; mainly the lexer was tossing chars
+ with the hi-order bit set. Not nice. Also, allow @ in extension
+ names, and a backslash, also.
+
+2007-11-27 17:01 +0000 [r89637] Joshua Colp <jcolp@digium.com>
+
+ * main/utils.c: Ensure the value returned from ast_random is
+ between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348)
+ Reported by: sperreault
+
+2007-11-27 16:13 +0000 [r89635] Russell Bryant <russell@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 89634 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) |
+ 3 lines Add a note to the sample voicemail config noting that
+ when using IMAP storage, only the first format specified will be
+ attached to the message. ........
+
+2007-11-27 15:41 +0000 [r89632] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007)
+ | 3 lines Default result of STAT should be "0" not "". Reported
+ via the -users mailing list, fixed by me. ........
+
+2007-11-27 07:36 +0000 [r89625] Olle Johansson <oej@edvina.net>
+
+ * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov
+ 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304)
+ Reported by: pj ........
+
+2007-11-27 06:47 +0000 [r89623] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample,
+ include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1
+ line closes issue #11379; OK, this is an attempt to make both
+ sides happy. To the cdr.conf file, I added the option
+ 'unanswered', which defaults to 'no'. In this mode, you will see
+ a cdr for a call, whether it was answered or not. The disposition
+ will be NO ANSWER or ANSWERED, as appropriate. The src is as
+ you'd expect, the destination channel will be one of the channels
+ from the Dial() call, usually the last in the list if more than
+ one chan was specified. With unanswered set to 'yes', you will
+ still see this cdr entry in both cases. But in the case where the
+ dial timed out, you will also see a cdr for each line attempted,
+ marked NO ANSWER, with no destination channel name. The new
+ option defaults to 'no', so you don't see the pesky extra cdr's
+ by default, and you will not see the irritating 'not posted'
+ messages. ........
+
+2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson <mmichelson@digium.com>
+
+ * pbx/ael/ael-test/ael-test19/extensions.ael,
+ pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt,
+ pbx/ael/ael-test/ael-test3/extensions.ael,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances
+ of "CALLERID(number)" to "CALLERID(num)" for consistency's sake
+ (closes issue #11381, reported and patched by jon)
+
+ * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov
+ 2007) | 7 lines After issuing a "say load new", if a caller hangs
+ up during the middle of playback of a number, app_playback will
+ continue to try to play the remaining files. With this change, no
+ more files will be played back upon hangup. (closes issue #11345,
+ reported and patched by IgorG) ........
+
+ * /: Blocked revisions 89616 via svnmerge ........ r89616 |
+ mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5
+ lines After issuing a "say load new" tons of warning messages are
+ printed out to the CLI every time do_say in app_playback is
+ called. Removing these warnings ........
+
+2007-11-26 22:52 +0000 [r89615] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Update the configure script check for
+ libpri to check for the newest function that was just added.
+ Cresl1n, please keep this in mind when making these changes to
+ libpri or libss7.
+
+2007-11-26 21:23 +0000 [r89613] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Rename
+ "limitonpeer" to "counteronpeer" since the call-limit is
+ deprecated. Both still works in this version.
+
+2007-11-26 21:14 +0000 [r89612] Joshua Colp <jcolp@digium.com>
+
+ * main/dial.c, /: Merged revisions 89610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2
+ lines Fix issues with async dialing with an application
+ executing. The application has to be terminated and control
+ returned to the thread before hanging things up. (issue #BE-252)
+ ........
+
+2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting, doxygenification
+
+ * channels/chan_sip.c: Formatting changes, cleaning up some code
+
+ * include/asterisk/doxyref.h, channels/chan_sip.c: Start using
+ Doxygen groupings to group variables and defines.
+
+ * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark
+ "concise" as deprecated - Restructure other changes to
+ UPGRADE.txt and CHANGES We're still looking for scripts that
+ replace asterisk -rx "show shannels concise" by using the manager
+ interface, but still produces the same output. Anyone?
+
+2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c, apps/app_queue.c: Perform some module use
+ counting audits. This is now done outside the scope of the
+ application/dialplan function so they do not need to worry about
+ it.
+
+ * /, res/res_features.c: Merged revisions 89599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6
+ lines Add module counting removal for error conditions. (closes
+ issue #11333) Reported by: Laureano Patches:
+ res_features_v2.c.patch uploaded by Laureano (license 265)
+ ........
+
+2007-11-26 17:49 +0000 [r89596] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 89594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) |
+ 3 lines Add channel locking to a function that needed to be doing
+ it. This is just a little something I noticed while working on a
+ completely unrelated issue. ........
+
+2007-11-26 17:46 +0000 [r89595] Steve Murphy <murf@digium.com>
+
+ * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes
+ issue #11341; made changes to make utils again right with the
+ MTX_PROFILE world.
+
+2007-11-26 17:38 +0000 [r89593] Joshua Colp <jcolp@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6
+ lines Use ast_free to free memory, or else we shall implode if
+ MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys
+ Patches: pbx.pbx_config.c.diff uploaded by ys (license 281)
+ ........
+
+2007-11-26 17:26 +0000 [r89591] Steve Murphy <murf@digium.com>
+
+ * main/hashtab.c: closes issue #11356; Many thanks to snuffy for
+ his code review and changes to cut down duplication. I tested
+ this against hashtest, and it passes. I reviewed the changes, and
+ they look reasonable. I had to remove a few const decls to make
+ things compile on my workstation,
+
+2007-11-26 17:25 +0000 [r89590] Russell Bryant <russell@digium.com>
+
+ * Makefile: make sure we check to see if the configure script has
+ been executed on a new checkout or after a distclean
+
+2007-11-26 17:23 +0000 [r89589] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov
+ 2007) | 6 lines Close the audio file before sending it to the
+ post processing application. (closes issue #11357) Reported by:
+ reformed Patches: mixmonitor.patch uploaded by reformed (license
+ 330) ........
+
+2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007)
+ | 2 lines when parsing application options that take arguments,
+ don't indicate that the option was supplied unless a
+ non-zero-length argument was found for it ........
+
+2007-11-26 16:24 +0000 [r89583] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to
+ pnlarsson for noting the spelling error in the cli commands.
+ Also, added some verbage about the new algorithm to CHANGES.
+
+2007-11-26 16:20 +0000 [r89582] Joshua Colp <jcolp@digium.com>
+
+ * main/utils.c: Revert change for 11348 until it can be looked at
+ even more.
+
+2007-11-26 15:50 +0000 [r89581] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov
+ 2007) | 6 lines Revert vmu->email back to an empty string if it
+ was empty when imap_store_file was called. This prevents sending
+ a duplicate e-mail. (closes issue #11204, reported by spditner,
+ patched by me) ........
+
+2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 89577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6
+ lines If channel allocation fails because the alert pipe could
+ not be created also free the scheduler context. (closes issue
+ #11355) Reported by: eliel Patches: main.channel.c.patch uploaded
+ by eliel (license 64) ........
+
+ * main/utils.c: Make the behavior of using /dev/urandom for random
+ numbers the same as random(). (closes issue #11348) Reported by:
+ sperreault Patches: ast_random2.diff uploaded by sperreault
+ (license 252)
+
+ * channels/chan_sip.c: Instead of printing out one codec in sip
+ show channels print out all of the native ones (this is for
+ video). (closes issue #11366) Reported by: ovi
+
+ * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4
+ lines When unloading app_meetme destroy any auto created contexts
+ created by SLA. (closes issue #11367) Reported by: eliel ........
+
+ * apps/app_controlplayback.c: Don't crash if the 'o' option of
+ ControlPlayback is used without any value. (closes issue #11375)
+ Reported by: johan
+
+2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_usbradio.c: Formatting changes
+
+ * main/channel.c, include/asterisk/channel.h: Try to get channel.h
+ and channel.c aligned in regards to ast_set_callerid as well as
+ change name of variables to follow the rest of the naming.
+
+2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, res/res_config_odbc.c, /,
+ res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
+ 89559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007)
+ | 14 lines We previously attempted to use the ESCAPE clause to
+ set the escape delimiter to a backslash. Unfortunately, this does
+ not universally work on all databases, since on databases which
+ natively use the backslash as a delimiter, the backslash itself
+ needs to be delimited, but on other databases that have no
+ delimiter, backslashing the backslash causes an error. So the
+ only solution that I can come up with is to create an option in
+ res_odbc that explicitly specifies whether or not backslash is a
+ native delimiter. If it is, we use it natively; if not, we use
+ the ESCAPE clause to make it one. Reported by: elguero Patch by:
+ tilghman (Closes issue #11364) ........
+
+ * channels/chan_sip.c: Typo (someone needs to test compile before
+ committing his changes)
+
+2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: More doxygen changes
+
+ * channels/chan_sip.c: Housekeeping
+
+ * channels/chan_sip.c: Formatting, doxygen updates
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: -
+ Deprecate "call-limit" in chan_sip. No other channel driver
+ enforces call-limits and we now have the groupcount system to
+ implement call-limits in the dialplan. You can use the "setvar"
+ option in realtime/sip.conf to set limits per device. - Implement
+ "callcounter" as a new option to enable the call counting we need
+ to report device status to queue, manager and SIP subscriptions.
+ The call counter setting is now enabled in the code by setting
+ the device call-limit to 999. When we remove the call limit, we
+ can simply enable this with a boolean setting.
+
+ * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping...
+ - Fix typo in chan_sip - Remove changes to caller ID structure,
+ moving it to branch (russellb)
+
+2007-11-24 21:00 +0000 [r89547] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c,
+ configs/extensions.conf.sample: closes issue #11363; where the
+ pattern _20x. buried in an included context, didn't match 2012;
+ There were a small set of problems to fix: 1. I needed NOT to
+ score patterns unless you are at the end of the data string. 2.
+ Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize
+ the patterns in the trie to caps. 3. When a pattern ends with dot
+ or exclamation, CANMATCH/MATCHMORE should always report this
+ pattern, no matter the length. With this commit, I also supplied
+ the wish of Luigi, where the user can select which pattern
+ matching algorithm to use, the old (legacy) pattern matcher, or
+ the new, trie based matcher. The OLD matcher is the default. A
+ new [general] section variable, extenpatternmatchnew, is added to
+ the extensions.conf, and the example config has it set to false.
+ If true, the new matcher is used. In all other respects, the
+ context/exten structs are the same; the tries and hashtabs are
+ formed, but in the new mode the tries are not used. A new CLI
+ command 'dialplan set extenpatternmatch true/false' is provided
+ to allow switching at run time. I beg users that are forced to
+ return to the old matcher to please report the reason in the bug
+ tracker. Measured the speed benefit of the new matcher against an
+ impossibly large context with 10,000 extensions: the new matcher
+ is 374 times faster.
+
+2007-11-24 17:07 +0000 [r89546] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007)
+ | 5 lines Free some frames that would otherwise leak on error.
+ Reported by: Laureano Patch by: Laureano,tilghman (Closes issue
+ #11351) ........
+
+2007-11-24 16:53 +0000 [r89544] Steve Murphy <murf@digium.com>
+
+ * main/app.c: Added <sys/file.h> include to allow trunk to compile.
+ Hope this doesn't louse thing up.
+
+2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_h323.c: remove a DEBUG_THREADS message that
+ accesses private lock fields. If needed, the code to extract this
+ information should be implemented in some generic header or
+ library and the function called here. (closed bug #11362)
+
+ * main/acl.c, main/http.c, main/app.c: remove some unnecessary
+ includes
+
+2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007)
+ | 9 lines Currently, zero-length voicemail messages cause a
+ hangup in VoicemailMain. This change fixes the problem, with a
+ multi-faceted approach. First, we do our best to avoid these
+ messages from being created in the first place, and second, if
+ that fails, we detect when the voicemail message is zero-length
+ and avoid exiting at that point. Reported by: dtyoo Patch by:
+ gkloepfer,tilghman (Closes issue #11083) ........
+
+ * main/manager.c, /: Merged revisions 89536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007)
+ | 10 lines Up until this point, the XML output of the manager has
+ been technically invalid, due to the repetition of certain
+ parameters in a single event. This caused various issues for XML
+ parsers, some of which refused to parse at all, given the
+ invalidity of the rendered XML. So this commit fixes the XML
+ output, ensuring that each entity parameter has a unique name,
+ thus ensuring valid XML. Reported by: msetim Patch by: tilghman
+ (Closes issue #10220) ........
+
+ * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23
+ Nov 2007) | 5 lines Use ESCAPE clause for the first parameter,
+ not just 2nd-Nth parameters. Reported by: apsaras Patch by:
+ tilghman (Closes issue #11353) ........
+
+2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_oss.c: put in the necessary hooks for video support
+ in the console. This is a NOP as far as the current code is
+ concerned, but there is already support in ./configure and the
+ Makefiles for the various libraries used by console_video.c (not
+ yet in the tree) so addition is trivial.
+
+ * channels/chan_sip.c: set rtpmap video info according to what is
+ read from SDP; make the format explicit in a debug message; print
+ the audio instead of aggregated peer capability in a debugging
+ msg.
+
+2007-11-23 09:40 +0000 [r89531] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/channel.h: Let's start with implementing the
+ base architecture for UTF8 caller ID's so we can handle multiple
+ formats properly. This is not carved in stone, but a proposal to
+ start with. We need to add support for transliterations as well
+ as UTF8 handling, propably with libiconv. Murf is looking into
+ that for the dialplan.
+
+2007-11-23 09:03 +0000 [r89530] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/image.h, formats/format_jpeg.c: formatting
+ cleanup on the header, normalization of the assignment of
+ descriptor fields.
+
+2007-11-23 02:37 +0000 [r89529] Russell Bryant <russell@digium.com>
+
+ * configs/agents.conf.sample, /: Merged revisions 89527 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) |
+ 3 lines mvanbaak pointed out a spelling error in this sample
+ configuration file. While I was at it, I went ahead and tweaked
+ it a little bit more. ........
+
+2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo <rizzo@icir.org>
+
+ * doc/CODING-GUIDELINES: new info on the management of headers
+
+ * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c,
+ apps/app_milliwatt.c: more header removal
+
+ * include/asterisk/channel.h: formatting cleanup
+
+ * include/asterisk.h, apps/app_read.c, apps/app_record.c,
+ apps/app_echo.c, apps/app_readexten.c,
+ include/asterisk/channel.h, apps/app_system.c,
+ apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h,
+ apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c,
+ include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c,
+ apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c,
+ apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a
+ little bit the content of header files to reduce dependencies. In
+ this commit: - move the ast_register/unregister_app functions to
+ module.h to avoid the need to include pbx.h for the simpler apps;
+ - move the ast_group structure to channel.h to remove the
+ dependency of app.h on linkedlists.h Note, this is a long process
+ that I am doing in small steps. The main difficulty is that now
+ for each subsystem we have a single header (e.g. channel.h)
+ included by the subsystem provider (usually one file, e.g.
+ channel.c) and by its clients (dozens of them, e.g. we have some
+ 70+ apps and 30+ functions). This requires the clients to include
+ all the extra headers required by the provider (eg. lock.h,
+ linkedlists.h, definitions of substructures...) even though many
+ of the clients would be just happy with opaque struct
+ declarations and function prototypes. The long term plan is to
+ eventually rectify this structure so that the compilation can
+ become faster, and also APIs are more stable.
+
+ * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c,
+ apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove
+ some useless includes
+
+ * include/asterisk/audiohook.h, apps/app_dictate.c,
+ apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c,
+ apps/app_mixmonitor.c, apps/app_stack.c,
+ apps/app_controlplayback.c: more removal of redundant headers
+
+ * apps/app_read.c, apps/app_echo.c, apps/app_record.c,
+ apps/app_userevent.c, apps/app_image.c, apps/app_system.c,
+ apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c,
+ apps/app_while.c: remove redundant headers
+
+ * main/file.c, main/netsock.c: more removal of fcntl.h and other
+ system headers
+
+ * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
+ codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c,
+ res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c,
+ formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c,
+ apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c,
+ codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which
+ are either useless or done elsewhere
+
+ * formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c, include/asterisk/_private.h,
+ formats/format_wav_gsm.c, formats/format_ilbc.c,
+ include/asterisk/file.h, formats/format_vox.c,
+ formats/format_pcm.c, main/file.c, formats/format_h263.c,
+ formats/format_g723.c, formats/format_h264.c,
+ include/asterisk/frame.h, formats/format_jpeg.c,
+ formats/format_g726.c, formats/format_gsm.c,
+ formats/format_g729.c: implement the split of file.h and
+ mod_format.h
+
+ * include/asterisk/mod_format.h (added): Add a specific header for
+ providers of file and format handling routines, moving here
+ structs and function declarations formerly in file.h
+
+2007-11-21 23:54 +0000 [r89513] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c,
+ res/res_features.c, apps/app_queue.c, channels/chan_iax2.c:
+ closes issue #11285, where an unload of a module that creates a
+ dialplan context, causes a crash when you do a 'dialplan show' of
+ that context. This is because the registrar string is defined in
+ the module, and the stale pointer is traversed. The reporter
+ offered a patch that would always strdup the registrar string,
+ which is practical, but I preferred to destroy the created
+ contexts in each module where one is created. That seemed more
+ symmetric. There were only 6 place in asterisk where this is
+ done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial,
+ and app_queue. The two apps destroyed the context, but left the
+ contexts. All is fixed now and unloads should be dialplan
+ friendly.
+
+2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo <rizzo@icir.org>
+
+ * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c,
+ channels/chan_local.c, apps/app_record.c, funcs/func_strings.c,
+ apps/app_sayunixtime.c, apps/app_test.c,
+ apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
+ apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c,
+ channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c,
+ pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c,
+ apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c,
+ channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c,
+ apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c,
+ funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c,
+ channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c,
+ channels/iax2-provision.c, funcs/func_cut.c,
+ apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
+ apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c,
+ apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
+ channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
+ apps/app_voicemail.c, channels/chan_unistim.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c,
+ apps/app_readexten.c, funcs/func_vmcount.c,
+ channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c,
+ cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c,
+ channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c,
+ funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
+ apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c,
+ funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c,
+ apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
+ apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c,
+ apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
+ channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ apps/app_channelredirect.c, apps/app_flash.c,
+ apps/app_directed_pickup.c, funcs/func_blacklist.c,
+ channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c,
+ channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c,
+ apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c:
+ remove another set of redundant #include "asterisk/options.h"
+
+ * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c,
+ main/say.c, res/res_features.c, main/devicestate.c, main/utils.c,
+ res/res_musiconhold.c, res/res_jabber.c, main/indications.c,
+ main/enum.c, res/res_config_sqlite.c, main/config.c,
+ main/loader.c, main/term.c, main/cli.c, main/io.c,
+ main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c,
+ res/res_config_odbc.c, main/manager.c, res/res_agi.c,
+ main/http.c, main/logger.c, res/res_realtime.c, main/app.c,
+ main/image.c, main/dns.c, main/db.c, res/res_speech.c,
+ main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c,
+ main/translate.c, res/res_crypto.c, res/res_adsi.c,
+ main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c,
+ res/res_ael_share.c, res/res_monitor.c, main/rtp.c,
+ main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c,
+ main/adsistub.c, main/abstract_jb.c, main/file.c,
+ main/callerid.c, main/astmm.c, main/audiohook.c,
+ formats/format_g726.c, main/asterisk.c, res/res_odbc.c,
+ main/dsp.c: remove a bunch of useless #include "options.h"
+
+2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Remove unneccessary explicit case for BRI
+
+ * channels/chan_zap.c: Take some debug code out :-)
+
+2007-11-21 22:20 +0000 [r89508] Luigi Rizzo <rizzo@icir.org>
+
+ * main/cygload.c: add a missing return
+
+2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add BRI support to chan_zap
+
+2007-11-21 21:30 +0000 [r89506] Luigi Rizzo <rizzo@icir.org>
+
+ * utils/Makefile, configure, configure.ac: enable support for stack
+ backtrace for stuff built in utils/ (this was present in the main
+ tree but forgotten here).
+
+2007-11-21 20:38 +0000 [r89505] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: closes issue #11290; the proposed patch was a good
+ guess, and would solve the bug to some extent, but was really
+ masking the real issue, that there were bad entries in the table.
+ This fix removes the condition that the hashtab updates be done
+ on exten removal only when the pattern_tree was present, which is
+ silly. The operations that apply to the pattern tree are instead
+ made conditional. Also, threw back in routines that kpfleming
+ deleted because of probs in the 64-bit world. Tested on both 32
+ and 64-bit machines (compile). Tested the reload problem with
+ over 20 reloads, and no problems. If you find more problems,
+ please reopen 11290.
+
+2007-11-21 20:22 +0000 [r89504] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Simplify comparison in parking fix
+
+2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov
+ 2007) | 3 lines Fix a small error I made in my previous commit
+ ........
+
+ * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov
+ 2007) | 5 lines Changing an inaccurate debug message to be less
+ inaccurate. Under the circumstances, this message would always
+ report that there were 0 members available, even though that may
+ not be true. ........
+
+2007-11-21 19:20 +0000 [r89492] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_features.c: Merged revisions 89491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) |
+ 4 lines If a channel gets masqueraded in the middle of a park,
+ don't play the announcement to the masqueraded channel, and dial
+ back to the original channel on timeout. ........
+
+2007-11-21 18:52 +0000 [r89490] Russell Bryant <russell@digium.com>
+
+ * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove
+ the FAX_DETECT define and only do the calculations if fax
+ detection is enabled on the dsp. (closes issue #11331) Reported
+ by: dimas Patches: dsp.patch uploaded by dimas (license 88)
+
+2007-11-21 18:38 +0000 [r89489] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set
+ READSTATUS as an indication of the result Also, some cleanup to
+ CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman
+ (Closes issue #11004)
+
+2007-11-21 18:24 +0000 [r89488] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: fix a small gramatical error in a comment
+
+2007-11-21 18:19 +0000 [r89487] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: There existed about a 1 in 4 billion chance that
+ reading from /dev/urandom would return LONG_MIN (1 in 9
+ quintillion if using 64-bit longs). Since there is no positive
+ equivalent of LONG_MIN, the result of labs() in this case is
+ unpredictable. This fixes that situation. (closes issue #11336,
+ reported and patched by sperreault)
+
+2007-11-21 16:24 +0000 [r89484] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c: Fix some code that was supposed to
+ ensure that a buffer was terminated, but was writing to the wrong
+ byte. Also, remove some non-thread safe test code. (closes issue
+ #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by
+ IgorG (license 20) - additional changes by me
+
+2007-11-21 16:08 +0000 [r89483] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I
+ attempted to port to trunk as well. Unfortunately, since trunk
+ uses read/write locks for the context lock, it means that I have
+ actually *introduced* a deadlock condition since they are not
+ recursive. Removing this change for now and will look into
+ introducing a different one.
+
+2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c,
+ utils/conf2ael.c: move these forward declarations back to
+ asterisk.h where they belong... even though asterisk.h includes
+ compat.h, these declarations have nothing to do with the being
+ platform-compatible and are directly related to being part of
+ Asterisk
+
+ * channels/chan_usbradio.c: get this to actually compile...
+
+ * main/pbx.c: remove some debugging code that doesn't compile on
+ 64-bit platforms
+
+2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: OOOps! All the debug stuff I inserted was
+ accidentally committed. I hereby revert it.
+
+ * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks
+ to snuffy for his work on neatening up the code and removing
+ duplicated code.
+
+2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/gentone-ulaw.c (removed): remove this file, it is not
+ used anywhere.
+
+ * main/astmm.c: add missing paths.h
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: add
+ check for video4linux
+
+2007-11-21 01:09 +0000 [r89474] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: A free in add_pri was ultimately the source of the
+ grief we were having with parking. This set of changes fixes that
+ problem, and introduces some more error messages, and puts debugs
+ into ifdefs for what could be short-term usage. Txs to Terry W.
+ for his help, guidance, and especially patience.
+
+2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo <rizzo@icir.org>
+
+ * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c,
+ main/minimime/mm.h, utils/check_expr.c: more header
+ removal/normalization
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: X11
+ checks (at least some - for other platforms with unusual X11
+ locations you might need to add more directories)
+
+2007-11-21 00:21 +0000 [r89470] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c, CHANGES: Merge changes from
+ team/russell/sla_trunk_moh ... * Added the ability to specify the
+ music on hold class used to play into the conference when there
+ is only one member and the M option is used. * Added the ability
+ to specify a music on hold class to play instead of ringing for
+ the SLATrunk application. (patched by me, and tested internally)
+
+2007-11-21 00:20 +0000 [r89469] Luigi Rizzo <rizzo@icir.org>
+
+ * makeopts.in: complete support for X11
+
+2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk
+ build again
+
+ * main/say.c: Add support for new recorded character sounds Closes
+ issue #5208
+
+2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
+ apps/app_dictate.c, apps/app_test.c, apps/app_ices.c,
+ apps/app_followme.c, channels/chan_iax2.c, main/config.c,
+ main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c,
+ main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h,
+ res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c,
+ main/app.c, main/image.c, apps/app_directory.c, main/db.c,
+ cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c,
+ include/asterisk/utils.h, include/asterisk.h, main/pbx.c,
+ channels/chan_sip.c, res/res_crypto.c,
+ include/asterisk/channel.h, res/res_monitor.c,
+ include/asterisk/paths.h, main/file.c, apps/app_sms.c,
+ include/asterisk/ael_structs.h, pbx/pbx_config.c,
+ apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
+ move asterisk/paths.h outside asterisk.h and into those files who
+ really need it.
+
+ * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c,
+ main/threadstorage.c, main/devicestate.c,
+ include/asterisk/_private.h (added), main/astobj2.c,
+ main/loader.c, main/term.c, main/cli.c, main/channel.c,
+ main/manager.c, main/logger.c, build_tools/strip_nonapi,
+ main/event.c, main/asterisk.c, main/db.c: move internal function
+ declarations to include/asterisk/_private.h
+
+2007-11-20 19:29 +0000 [r89464] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: i got a little carried away with commas
+ ...
+
+2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/module.h, build_tools/make_buildopts_h,
+ main/loader.c: switch compile-time option checking to string
+ storage mode in this branch too
+
+2007-11-20 19:11 +0000 [r89460] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: fix the zaptel configure script check
+
+2007-11-20 18:20 +0000 [r89459] Luigi Rizzo <rizzo@icir.org>
+
+ * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before
+ regenerating configure, i have more changes)
+
+2007-11-20 17:59 +0000 [r89458] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 89457 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov
+ 2007) | 9 lines According to comments in main/pbx.c, it is
+ essential that if we are going to lock the conlock as well as the
+ hints lock, it must be locked in that respective order. In order
+ to prevent a potential deadlock, we need to lock the conlock
+ prior to locking the hints lock in ast_hint_state_changed (see
+ the call stack example on issue #11323 for how this can happen).
+ (closes issue #11323, reported by eelcob, suggestion for patch by
+ eelcob, patch by me) ........
+
+2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo <rizzo@icir.org>
+
+ * makeopts.in: prepare to support console_video
+
+ * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile,
+ channels/Makefile: Fix building of modules under cygwin. After
+ this commit we can actually load modules under windows, and we
+ can start debugging more interesting problems related to the load
+ order and functionality of modules.
+
+2007-11-20 16:11 +0000 [r89453] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Changed occurrences of "busy-level" to
+ "busylevel" in sip.conf.sample in light of commit 89441. Thanks
+ to pj for pointing out the need for this (closes issue #11307,
+ reported by pj)
+
+2007-11-20 15:39 +0000 [r89452] Luigi Rizzo <rizzo@icir.org>
+
+ * configure, configure.ac, acinclude.m4: add an argument for extra
+ headers to AC_EXT_LIB_CHECK, and on passing simplify the code.
+ Too bad that every time we need to regenerate configure...
+
+2007-11-20 15:30 +0000 [r89451] Steve Murphy <murf@digium.com>
+
+ * /, doc/tex/queues-with-callback-members.tex: Merged revisions
+ 89450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1
+ line closes issue #11324; break statements missing in switch
+ cases. ........
+
+2007-11-20 15:00 +0000 [r89449] Joshua Colp <jcolp@digium.com>
+
+ * main/translate.c: Minor documentation tweak and if an incorrect
+ parameter is given to core show translation return the usage
+ information. (closes issue #11316) Reported by: eliel Patches:
+ translate.c.patch uploaded by eliel (license 64)
+
+2007-11-20 14:54 +0000 [r89448] Luigi Rizzo <rizzo@icir.org>
+
+ * configure, acinclude.m4: comment a bit the code in acinclude.m4
+ There is still a lot of code to clean up there, but hopefully
+ this should clarify what goes on in there.
+
+2007-11-20 14:49 +0000 [r89447] Joshua Colp <jcolp@digium.com>
+
+ * channels/h323/ast_h323.cxx: Include the compatibility header file
+ in ast_h323.cxx for compatibility reasons. (closes issue #11311)
+ Reported by: falves11
+
+2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Fix sip show history. Closes issue #11312
+
+ * channels/chan_sip.c: Change terminology a bit for CLI commands
+ handling SIP channels/calls/dialogs/whatever. Closes issue #11312
+
+2007-11-20 07:42 +0000 [r89443] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile
+ changes to build loadable modules under cygwin (not complete yet
+ - still need to sort out dependecies on res_*)
+
+2007-11-20 00:17 +0000 [r89442] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Get rid of some debug messages in pbx.c
+
+2007-11-19 23:24 +0000 [r89441] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in
+ sip.conf to "busylevel" to be more parallel with the SIPPEER()
+ argument of the same name. The deprecation procedure is not being
+ used here since this is a trunk-only option. (closes issue
+ #11307, reported by pj, patched by me)
+
+2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/module.h: Be a bit more pedantic about the type
+ for holding the md5 sum for the build options. Also, doxygenify
+ the comment.
+
+ * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect
+ which options were compiled in
+
+2007-11-19 22:55 +0000 [r89438] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: These changes were made in response to
+ niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in
+ two different contexts, included in the same context. In that
+ particular case, we were behaving the same as 1.4, but after
+ experimenting, I quickly found that if 20 and 201 were in the
+ same extension, 1.4 would return 201, and this code returns 20.
+ These changes now enable the current code to replicate the
+ behavior of 1.4 in respect to MATCHMORE in cases like this.
+
+2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c:
+ another few errno.h removals
+
+ * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
+ apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c,
+ pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
+ apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c,
+ apps/app_milliwatt.c: more errno.h removal
+
+ * funcs/func_sysinfo.c: remove unnecessary headers
+
+ * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary
+ includes.
+
+2007-11-19 20:13 +0000 [r89429] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Change delimiter of SIPPEER to be comma
+ (instead of pipe) and further deprecate the old ':' delimiter
+ Reported by: pj Patch by: tilghman Closes issue #11305
+
+2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo <rizzo@icir.org>
+
+ * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c,
+ codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c,
+ codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c,
+ codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless
+ includes from codecs
+
+ * formats/format_ilbc.c, formats/format_sln.c,
+ formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
+ formats/format_ogg_vorbis.c, formats/format_g723.c,
+ formats/format_h263.c, formats/format_h264.c,
+ formats/format_wav_gsm.c, formats/format_g726.c,
+ formats/format_jpeg.c, formats/format_gsm.c,
+ formats/format_g729.c: format handlers don't need network, lock,
+ channel and scheduler headers
+
+ * include/asterisk.h, include/asterisk/compat.h,
+ include/asterisk/lock.h, utils/extconf.c,
+ include/asterisk/abstract_jb.h: move the declaration of struct
+ ast_channel ast_frame and ast_module to compat.h so it is always
+ available - hopefully this will let us reduce the number of
+ inclusions of channel.h and frame.h
+
+ * main/udptl.c, main/autoservice.c, funcs/func_rand.c,
+ cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c,
+ main/threadstorage.c, main/say.c, funcs/func_env.c,
+ funcs/func_strings.c, main/devicestate.c,
+ cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c,
+ main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c,
+ main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c,
+ main/sched.c, main/pbx.c, funcs/func_timeout.c,
+ funcs/func_math.c, funcs/func_cut.c, main/chanvars.c,
+ main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c,
+ funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c,
+ funcs/func_iconv.c, main/alaw.c, main/asterisk.c,
+ funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c,
+ main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c,
+ funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c,
+ funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c,
+ main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c,
+ funcs/func_groupcount.c, main/manager.c, main/tdd.c,
+ funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c,
+ main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c,
+ funcs/func_db.c, main/dnsmgr.c, main/translate.c,
+ main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c,
+ cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c,
+ funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c,
+ main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c,
+ funcs/func_callerid.c, main/dsp.c: another bunch of include
+ removals (errno.h and asterisk/logger.h)
+
+ * channels/chan_local.c, apps/app_record.c,
+ apps/app_alarmreceiver.c, apps/app_chanisavail.c,
+ apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c,
+ channels/chan_skinny.c, formats/format_pcm.c,
+ apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c,
+ codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c,
+ codecs/codec_g726.c, formats/format_ogg_vorbis.c,
+ apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c,
+ apps/app_speech_utils.c, apps/app_sendtext.c,
+ formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c,
+ apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c,
+ res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c,
+ apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c,
+ channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c,
+ res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c,
+ apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c,
+ apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ apps/app_adsiprog.c, codecs/codec_lpc10.c,
+ res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c,
+ codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c,
+ channels/chan_jingle.c, apps/app_channelredirect.c,
+ apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c,
+ formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c,
+ apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c,
+ apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c,
+ formats/format_wav_gsm.c, res/res_smdi.c,
+ include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c,
+ channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
+ formats/format_jpeg.c, formats/format_gsm.c,
+ apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
+ apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c,
+ channels/chan_agent.c, apps/app_disa.c,
+ channels/iax2-provision.c, res/res_ael_share.c,
+ apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c,
+ channels/chan_misdn.c, apps/app_waitforring.c,
+ apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c,
+ apps/app_zapateller.c, res/res_indications.c,
+ codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c,
+ apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c,
+ codecs/codec_zap.c, res/res_config_sqlite.c,
+ channels/misdn_config.c, apps/app_controlplayback.c,
+ formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c,
+ main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c,
+ res/res_clioriginate.c, apps/app_while.c, include/asterisk.h,
+ apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c,
+ res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
+ res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c,
+ apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c,
+ apps/app_flash.c, apps/app_directed_pickup.c,
+ channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
+ apps/app_senddtmf.c, formats/format_g729.c: include "logger.h"
+ and errno.h from asterisk.h - usage shows that they were included
+ almost everywhere. Remove some of the instances.
+
+2007-11-19 17:18 +0000 [r89422] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: a correction to code involved in an extension removal
+
+2007-11-19 16:29 +0000 [r89421] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan
+ function for retrieval of system information
+
+2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_features.c: Merged revisions 89419 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6
+ lines Print out the correct filename (features.conf) in the log
+ message when parkpos options are incorrect. (closes issue #11295)
+ Reported by: Laureano Patches: res_features.c.patch uploaded by
+ Laureano (license 265) ........
+
+ * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov
+ 2007) | 4 lines Clarify documentation a bit, include that a frame
+ has to pass through the core in order for the Local channel
+ optimization to happen. (closes issue #11246) Reported by: jon
+ ........
+
+2007-11-19 14:36 +0000 [r89412] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/logger.h: revert inclusion of options.h
+
+2007-11-19 14:03 +0000 [r89410] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_playback.c: Change warning messages (which are really
+ debug messages) into debug messages. (closes issue #11288)
+ Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded
+ by IgorG (license 20)
+
+2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Update CHANGES
+
+ * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan
+ function. With this, you can control the peer in the dialplan, so
+ you avoid placing outbound calls when the device has reached
+ busy-level. Reported by pj. Closes bug #11180
+
+ * main/acl.c: Add some debugging to the routines that finds our
+ local IP address. Related to bug #9225
+
+ * channels/chan_sip.c: Make some notes about a problem I found with
+ the OPTIONs handler while working with the bug tracker. Since we
+ don't authenticate devices (peers/users) on OPTIONS we don't have
+ the proper context set for the user/peer. However, we might not
+ want to process an authentication for every OPTIONS, so we could
+ have a config option for this, "optionsforceok" to always answer
+ 200 OK on the request and not check device or destination, nor
+ add a SDP. If Asterisk sends the OPTIONs request, it doesn't care
+ about the reply. Some devices use OPTIONs to discover
+ capabilities, since we should answer like an INVITE from the
+ device and we need to support that properly too, which we don't
+ today. So much to do :-)
+
+2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp <jcolp@digium.com>
+
+ * build_tools/make_buildopts_h: Add OSX into the logic that uses
+ md5 instead of md5sum.
+
+ * include/asterisk/compat.h: Use the easy way that rizzo mentioned,
+ only include malloc.h on the Windows platform.
+
+ * include/asterisk/compat.h: Revert last commit, apparently
+ buildbot lied to me.
+
+ * include/asterisk/compat.h: Change how we handle alloca to conform
+ with how it is suggested in the autoconf manual for
+ AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms
+ should be broken by this.
+
+ * configure, configure.ac: Change autoconf logic a bit so it says
+ what it is looking for in two instances where it didn't.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h, include/asterisk/network.h: Use autoconf
+ logic to determine the presence of
+ PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and
+ PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h
+ in "
+
+2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add SS7 Generic address support (#11156)
+
+2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/compat.h: if alloca.h is not present, try
+ malloc.h
+
+ * agi/Makefile: temporarily disable this target in mingw
+
+ * Makefile: will i ever get precedences for windows right ? in the
+ meantime, use a variable to ease enabling/disabling print
+ subdirectories.
+
+ * Makefile: reformulate dependencies in a more correct way
+
+2007-11-17 17:46 +0000 [r89388] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make
+ it so it will compile. Hope I did the right thing. And some
+ additions to removal of extens to take care of hashtab pointers
+ in all cases.
+
+2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile.moddir_rules, Makefile.rules: as discussed some time ago
+ on the -dev list, create embedde object with a .eo suffix even if
+ they are coming from .cc sources. This simplifies the handling in
+ the build scripts.
+
+ * include/asterisk/network.h: prefer socket.h over other variants
+ (winsock etc.)
+
+ * channels/chan_local.c, main/translate.c,
+ channels/chan_features.c, main/http.c, main/config.c: trim more
+ redundant headers
+
+ * main/acl.c: remove unnecessary includes
+
+ * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c,
+ main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by
+ previous mistake
+
+ * Makefile: wrong variable, wrong order -> broken build.
+
+ * include/asterisk/acl.h, include/asterisk/utils.h,
+ include/asterisk/autoconfig.h.in, include/asterisk/rtp.h,
+ configure.ac, main/acl.c, include/asterisk/netsock.h,
+ main/utils.c, include/asterisk/manager.h, main/netsock.c,
+ main/manager.c, res/res_agi.c, pbx/pbx_dundi.c,
+ include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
+ main/asterisk.c: start using asterisk/network.h for network
+ related headers. Also remove some unnecessary includes.
+
+ * include/asterisk/network.h (added): wrapper for all generic
+ network headers that have different names and locations on the
+ various systems.
+
+ * main/cygload.c: main is called main not amain!
+
+ * main/Makefile: conditional targets for building the windows
+ version
+
+ * Makefile: support cygwin targets
+
+ * Makefile.moddir_rules: and this is the last one to have asterisk
+ compile (not run yet) natively under cygwin.
+
+ * apps/app_sms.c: another cygwin compatibility fix. This one must
+ be handled in a better way in configure, also for other
+ architectures
+
+ * utils/Makefile, main/Makefile, utils/extconf.c: more
+ cygwin/mingw32 compatibility fixes
+
+ * include/asterisk/channel.h: use autoconf results to conditionally
+ compile timersub
+
+ * include/asterisk/lock.h: compatibility fixes for cygwin
+
+ * include/asterisk/compat.h: some version of flex produce code that
+ wants __STDC_VERSION__ defined, but the compiler does not always
+ define it.
+
+ * Makefile: these linker flags apply to both cygwin and mingw32
+
+ * utils/hashtest2.c: add a return NULL to a function that is
+ expected to return a value so compilers that don't understand
+ that this code is NOTREACHED will not complain (the fault is not
+ much on the compiler but on the declaration of pthread_exit on
+ certain platforms) s/certain platform/cygwin/ if you are really
+ curious
+
+ * main/loader.c: define RTLD_LOCAL for platforms that don't have
+ it. This is only to complete the build, clearly the linker
+ behaviour will be completely different and likely to cause
+ trouble in those cases.
+
+ * channels/Makefile: filter out modules that do not compile under
+ windows (this should be handled with the dependencies generated
+ by configure and menuselect, but will be fixed later)
+
+ * main/utils.c: netdb.h is used for gethostbyname, and it was not
+ included in some platforms.
+
+ * main/cygload.c (added): Loader for cygwin where asterisk is
+ really a big dll (something like this is already in 1.2)
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ timersub is a macro not a function, so write the check in a way
+ that detects both formats.
+
+2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_lua.c: fix the build of pbx_lua
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac, include/asterisk/io.h,
+ include/asterisk/channel.h: Update the configure script check for
+ sys/poll.h to also provide the result in
+ include/asterisk/autoconfig.h. Also, move the conditional include
+ of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h
+ instead of the two headers it existed in before.
+
+ * build_tools/make_buildopts_h: actually let this compile, oops :(
+
+ * build_tools/make_buildopts_h: Use the fix suggested by Tilghman
+ on the -dev to make cutting up the BUILDSUM friendly to non-bash
+ shells. I think this should work for BSD/mingw as well, but did
+ not yet remove the switch statement.
+
+2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: linker flags for mingw32
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: add
+ detection for timersub() and winsock.h/winsock2.h
+
+ * include/asterisk/endian.h: provide definitions for
+ __LITTLE_ENDIAN and __BIG_ENDIAN if not present.
+
+ * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h:
+ use poll as detected by configure
+
+ * configure, configure.ac, makeopts.in: use autoconf to check for
+ the existence of sys/poll.h
+
+ * build_tools/make_buildopts_h: this script is run on the build
+ system, not on the host.
+
+ * Makefile.moddir_rules: compatibility fix for mingw32
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4, makeopts.in: acinclude.m4: add a function to help
+ checking sdl-config, gtk-config and the like (this could be used
+ for gtk and gtk2 as well) Other files: add tests for sdl,
+ sdl_image and avcodec and regenerate configure and
+ autoconfig.h.in
+
+ * include/asterisk/autoconfig.h.in, configure.ac: add check for the
+ presence of glob
+
+ * channels/chan_jingle.c, channels/chan_unistim.c,
+ funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c,
+ utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
+ apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate
+ #include lines
+
+ * main/udptl.c, funcs/func_module.c, res/res_features.c,
+ funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c,
+ channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c,
+ pbx/pbx_loopback.c, funcs/func_blacklist.c,
+ channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c,
+ main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c:
+ remove a bunch of duplicate includes Reproduce with grep -r
+ #include . | grep -v .svn | grep -v Binary | sort | uniq -c |
+ sort -nr
+
+2007-11-16 23:44 +0000 [r89347] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Fix broken parking dial-back
+
+2007-11-16 23:33 +0000 [r89346] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: My goodness, haven't handled an extension deletion.
+ Add code to ast_context_remove_extension2() to remove an
+ extension from the trie. Done by marking it deleted. The
+ scoreboard won't update for it any more. Also, a couple of calls
+ to insert hashtab had a spurious ->exten, which was removed.
+
+2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/paths.h, include/asterisk.h: paths are already
+ in include/asterisk/paths.h so don't duplicate them in
+ include/asterisk.h
+
+ * include/asterisk/utils.h, include/asterisk/lock.h: whitespace
+ only change - adjust indentation and add some comments on the
+ content of these two files. utils.h (which is included in over
+ 150 files) contains a lot of unrelated functions which require
+ the inclusion of a large number of other headers. At some point
+ we should partition its content in a better way.
+
+2007-11-16 22:33 +0000 [r89340] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 89339 via svnmerge ........ r89339 | russell
+ | 2007-11-16 16:26:44 -0600 (Fri, 16 Nov 2007) | 5 lines
+ Temporarily revert revision 89325, which added md5 magic for
+ keeping track of what build options were used. We agreed that we
+ should remove this before making a 1.4 release, and then we can
+ put it back in. Then, we can take a month or so to play around
+ with it to get it how we want it. ........
+
+2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/logger.h: logger.h does not need options.h
+
+ * include/asterisk/utils.h, channels/chan_sip.c,
+ include/asterisk/astobj.h, include/asterisk/compat.h,
+ include/asterisk/channel.h, include/asterisk/strings.h,
+ utils/extconf.c, include/asterisk/frame.h,
+ include/asterisk/stringfields.h, include/asterisk/endian.h:
+ remove redundant #include "asterisk/compat.h", but make sure that
+ asterisk/compiler.h is included everywhere
+
+ * main/acl.c, main/asterisk.c: remove duplicate headers. Properly
+ check for netdb.h (there is actually tens of places to fix)
+
+ * Makefile.rules: put back default optimization to -O6 (previously
+ changed by mistake)
+
+ * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c,
+ apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c,
+ channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c,
+ apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c,
+ include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c,
+ main/pbx.c, funcs/func_timeout.c, formats/format_wav.c,
+ apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c,
+ apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c,
+ main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c,
+ main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h,
+ apps/app_voicemail.c, formats/format_sln.c,
+ apps/app_authenticate.c, apps/app_readexten.c,
+ apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules,
+ apps/app_setcallerid.c, include/asterisk/astmm.h,
+ res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c,
+ apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c,
+ main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c,
+ funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
+ res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c,
+ apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c,
+ main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c,
+ apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
+ main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c,
+ cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c,
+ formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c,
+ funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c,
+ main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c,
+ formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c,
+ main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c,
+ channels/iax2-provision.c, res/res_monitor.c, main/netsock.c,
+ apps/app_waitforring.c, main/fixedjitterbuf.c,
+ include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c,
+ channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c,
+ apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c,
+ res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c,
+ res/res_config_sqlite.c, main/fskmodem.c,
+ channels/misdn_config.c, apps/app_controlplayback.c,
+ cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c,
+ channels/chan_phone.c, funcs/func_enum.c, main/dial.c,
+ main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c,
+ main/logger.c, main/image.c, apps/app_ivrdemo.c,
+ res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c,
+ channels/chan_zap.c, main/slinfactory.c, res/res_convert.c,
+ pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c,
+ channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
+ channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c,
+ include/asterisk/strings.h, apps/app_senddtmf.c,
+ funcs/func_callerid.c, include/asterisk/time.h,
+ channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c,
+ apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c,
+ pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c,
+ main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c,
+ apps/app_morsecode.c, formats/format_ogg_vorbis.c,
+ channels/iax2-parser.c, apps/app_speech_utils.c,
+ include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c,
+ funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c,
+ utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c,
+ formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c,
+ channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
+ res/res_jabber.c, funcs/func_uri.c, main/io.c,
+ include/asterisk/abstract_jb.h, main/channel.c,
+ apps/app_minivm.c, res/res_realtime.c, main/dns.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c,
+ res/res_adsi.c, include/asterisk/plc.h,
+ apps/app_channelredirect.c, formats/format_vox.c,
+ main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c,
+ funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c,
+ cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c,
+ apps/app_sayunixtime.c, apps/app_test.c, main/config.c,
+ main/loader.c, main/term.c, main/cli.c, res/res_smdi.c,
+ include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
+ include/asterisk/utils.h, apps/app_privacy.c,
+ codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c,
+ funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c,
+ apps/app_transfer.c, include/asterisk/manager.h,
+ apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c,
+ channels/chan_misdn.c, funcs/func_curl.c,
+ channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c,
+ res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c,
+ include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c,
+ funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c,
+ include/asterisk/file.h, main/tdd.c, funcs/func_volume.c,
+ res/res_agi.c, main/app.c, apps/app_parkandannounce.c,
+ cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c,
+ res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
+ main/translate.c, include/asterisk/config.h, main/jitterbuf.c,
+ main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c,
+ funcs/func_extstate.c, apps/app_directed_pickup.c,
+ main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c,
+ main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c,
+ formats/format_g729.c: Start untangling header inclusion in a way
+ that does not affect build times - tested, there is no
+ measureable difference before and after this commit. In this
+ change: use asterisk/compat.h to include a small set of system
+ headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h,
+ stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the
+ inclusion is conditional on HAVE_FOO_H as determined by autoconf.
+ Normally, source files should not include any of the above system
+ headers, and instead use either "asterisk.h" or
+ "asterisk/compat.h" which does it better. For the time being I
+ have left alone second-level directories (main/db1-ast, etc.).
+
+2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Fixing a problem pointed out by Qwell
+
+ * main/manager.c: Added some locks that should have been around
+ astman_send_error, at least according to the comments. (closes
+ issue #11258, reported and patched by eliel)
+
+2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: This corrects a hashtab removal, given a bad argument
+
+ * main/pbx.c, res/res_features.c: This fixes a problem with pattern
+ ranges; and corrects a situation in res_features, where an
+ extension would be created with the name Zap/51, as an example.
+ THe / is bad because it would tend to mean that the 51 is to be
+ cid matched.
+
+2007-11-16 18:48 +0000 [r89328] Luigi Rizzo <rizzo@icir.org>
+
+ * build_tools/make_buildopts_h: both md5sum and variable
+ substitutions such as ${BUILDSUM:0:8} are not available in
+ FreeBSD. For the time being, put in a workaround so we can build
+ the system, and wait for the result of the discussion on whether
+ we can store the md5 as a string rather than 4 ints (if so, we
+ won't need more complex tricks with awk or sed for splitting the
+ md5). 1.4 will be fixed when we decide the issue.
+
+2007-11-16 17:11 +0000 [r89327] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Adding confirmation playback when
+ forwarding voicemail messages. This will attempt to play the
+ name(s) of the person(s) to whom you are forwarding the message
+ prior to prompting for prepending. If no name is found, the
+ extension is read back verbatim. (closes issue #9046, reported
+ and patched by jaroth)
+
+2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/module.h, build_tools/make_buildopts_h,
+ main/loader.c: Merged revisions 89325 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007)
+ | 4 lines To help combat problems where people build external
+ modules (asterisk-addons or others) and then change the build
+ options of the Asterisk build in a way that makes the
+ incompatible without warning, this commit introduces an MD5
+ signature of the important build-time options and includes that
+ signature into modules when they are built. When the loader loads
+ one of these modules and notices the problem, it will emit a
+ warning to console and refuse to initialize the module, as doing
+ so could cause the system to be unstable or even crash. If you
+ upgrade to this version of Asterisk, you must rebuild *all* of
+ your modules that came from other sources before trying to run
+ this version. If you are using Digium's G.729 binary codec
+ module, you will need v33 or newer. ........
+
+2007-11-16 15:44 +0000 [r89324] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov
+ 2007) | 5 lines Make realtime queues accessible from the
+ QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
+ patched by atis, with small modifications from me) ........
+
+2007-11-16 10:07 +0000 [r89322] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/config.h, main/config.c: add a small new
+ function to retrieve variables from a config once we have a
+ pointer to the category.
+
+2007-11-16 10:06 +0000 [r89321] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed #10631, about one way audio. thanks
+ IgorG again.
+
+2007-11-16 09:51 +0000 [r89320] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_oss.c: move the inner part of config file parsing
+ to a separate function, so it can be reused in the implementation
+ of cli commands when they have a similar syntax.
+
+2007-11-16 08:54 +0000 [r89319] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269,
+ thanks IgorG.
+
+2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher <tlesher@digium.com>
+
+ * main/utils.c, include/asterisk/stringfields.h: If we're going to
+ be passing a negative value for the size of a stringfield, in
+ order to indicate something, then using an UNSIGNED parameter is
+ bad, mmmmmkay?
+
+ * Makefile, /: Merged revisions 89302 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007)
+ | 2 lines Start Asterisk in Debian at a more reasonable time
+ (since zaptel is at level 20) ........
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15
+ Nov 2007) | 2 lines Fix an uninitialized memory read found by
+ valgrind ........
+
+ * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being
+ renamed.
+
+ * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007)
+ | 5 lines Yet another memory corruption issue. Reported by: atis
+ Patch by: tilghman Fixes issue #10923 ........
+
+2007-11-15 17:27 +0000 [r89297] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) |
+ 8 lines Update the SLAStation application to account for the case
+ where the SLA thread has a call out to the station, but the user
+ has pressed a line button to answer the call instead of picking
+ up the handset. If they do, the phone sends out a new INVITE. So,
+ the SLAStation app must check to see if it is picking up a
+ ringing trunk, and ensure that the other stations stop ringing.
+ (reported internally, patched by me, tested by mogorman) ........
+
+2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Get rid of a previously missed ast_log call for
+ debug, no longer nec.
+
+ * main/pbx.c: Perhaps I went overboard on initializing things. I
+ can remove unnecc. stuff later. A few bug fixes. Killing small
+ bugs on the way to killing bigger ones. Removed locking on
+ hashtabs; there's plenty of locks already being taken. A small
+ bug in the root_tree hashtab compare func.
+
+2007-11-15 16:20 +0000 [r89293] Luigi Rizzo <rizzo@icir.org>
+
+ * main/channel.c, apps/app_channelredirect.c, main/manager.c,
+ res/res_features.c, apps/app_softhangup.c,
+ include/asterisk/channel.h, include/asterisk/lock.h,
+ apps/app_senddtmf.c: access channel locks through
+ ast_channel_lock/unlock/trylock and not through ast_mutex
+ primitives. To detect all occurrences, I have renamed the lock
+ field in struct ast_channel so it is clear that it shouldn't be
+ used directly. There are some uses in res/res_features.c (see
+ details of the diff) that are error prone as they try and lock
+ two channels without caring about the order (or without
+ explaining why it is safe).
+
+2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp <jcolp@digium.com>
+
+ * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been
+ used, not 'increasing'.
+
+ * channels/chan_sip.c, channels/chan_h323.c,
+ channels/misdn_config.c: And file said... let trunk build again!
+ Accomplished by some more constification, and marking a function
+ in chan_sip as purposely unused until it is fixed up.
+
+2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 89288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov
+ 2007) | 3 lines Undoing previous commit since I realize it was
+ wrong ........
+
+ * main/manager.c, /: Merged revisions 89286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov
+ 2007) | 4 lines Adding a missing mutex unlock. (closes issue
+ 11256, reported and patched by ys) ........
+
+2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Always relying on the responses when
+ crossing NAT's are not a good solution, it breaks communication.
+ Rizzo - you need to implement a configuration option for this
+ code. It's good, but maybe should be off by default.
+
+ * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6
+ lines Don't send re-invites during pending INVITE transactions.
+ Patch by one47 - thanks! Closes issue #9305 ........
+
+ * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5
+ lines Improve support for multipart messages. Code by gasparz,
+ changes by me (mostly formatting). Thanks, gasparz! Closes issue
+ #10947 ........
+
+ * channels/chan_sip.c: Exit early instead of deciding to exit after
+ processing the message.
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add support for
+ application/dtmf SIP INFO dtmf handling. Yep, another way of
+ handling DTMF in SIP. Totally undocumented, but implemented in
+ enough devices so we have to support it. Code by sergee, small
+ changes by oej. Closes issue #11049
+
+2007-11-15 01:42 +0000 [r89277] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Had trouble playing with parking; spent a long time
+ trying to reason out MATCHMORE mode. made these updates and xfers
+ on zaptel lines seem to work ok now
+
+2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 89275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007)
+ | 5 lines When a recording ends with '#', we are improperly
+ trimming an extra 200ms from the recording. Reported by: sim
+ Patch by: tilghman Closes issue #11247 ........
+
+ * main/channel.c: Typo
+
+ * main/channel.c: Add callerid to the Hangup manager event.
+ Reported by: outtolunc Patch by: outtolunc Closes issue #11248
+
+2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Rescaled the weights of the patterns to give
+ something more independent of pattern length; and make . less
+ likely to win. Question: which should win for 14102241145--
+ _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will
+ win.
+
+ * main/pbx.c: A further problem highlighted by 11233 has been
+ resolved; a certain combination of patterns in a certain order,
+ led to a malformed trie, due to a ptr not being initialized in
+ the loop. Also, some tree printing prettifications.
+
+2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c,
+ res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more
+ typo in config.c; and missed conversions due to the constifying
+ of ast_variable_new parameters
+
+ * main/config.c: Typo
+
+2007-11-14 13:18 +0000 [r89268] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/acl.h, channels/chan_sip.c,
+ include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c,
+ main/acl.c, pbx/dundi-parser.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/enum.c, channels/chan_oss.c,
+ apps/app_playback.c, main/config.c, pbx/dundi-parser.h,
+ include/asterisk/abstract_jb.h, main/manager.c,
+ channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c,
+ main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c,
+ apps/app_voicemail.c: make the 'name' and 'value' fields in
+ ast_variable const char * This prevents modifying the strings in
+ the stored variables, and catched a few instances where this was
+ actually done. Given the differences between trunk and 1.4 (and
+ the fact that this is effectively an API change) it is better to
+ fix 1.4 independently. These are chan_sip.c::sip_register()
+ chan_skinny.c:: near line 2847 config.c:: near line 1774
+ logger.c::make_components() res_adsi.c:: near line 1049 I may
+ have missed some instances for modules that do not build here.
+
+2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant <russell@digium.com>
+
+ * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding
+ guidelines issues ... - handle memory allocation failures - add
+ an ast_ prefix to a publicly exported function - put curly braces
+ in the right places - add a bunch of spaces where they should be
+ be used
+
+ * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple
+ places - return errors from load_module / unload_module
+
+ * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the
+ syntax highlighting in my editor happy
+
+ * apps/app_queue.c: Instead of reserving 800 bytes for periodic
+ announcements, use an array of ast_str pointers and only alloate
+ space for the strings as needed.
+
+2007-11-14 01:16 +0000 [r89262] Joshua Colp <jcolp@digium.com>
+
+ * main/srv.c, /: Merged revisions 89260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4
+ lines Return the proper value when the srv_callback function
+ executes properly. (closes issue #11240) Reported by: jtodd
+ ........
+
+2007-11-14 01:15 +0000 [r89261] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Convert most of the strings in the call_queue
+ struct to use stringfields.
+
+2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, main/pbx.c: use simpler technique for removing
+ known entries from lists
+
+2007-11-14 00:33 +0000 [r89258] Russell Bryant <russell@digium.com>
+
+ * main/image.c: - Simplify removing an item from a list - move a
+ verbose message to after the item is added to the list - make use
+ of the ARRAY_LEN macro in one spot
+
+2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't
+ covered the case of a context with no patterns. (blush)
+
+ * main/pbx.c: closes issue #11233 -- where some fine points in the
+ algorithm to build the tree needed to be corrected. Many thanks
+ for the test case, jtodd
+
+2007-11-13 21:08 +0000 [r89255] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 89254 via svnmerge (closes issue #11238)
+ ........ r89254 | qwell | 2007-11-13 15:07:08 -0600 (Tue, 13 Nov
+ 2007) | 4 lines Fix building on newer systems which require a
+ third arg to open() when using O_CREAT. Issue 11238, reported by
+ puzzled. ........
+
+2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/lock.h: This fixes a build error on my mac. It
+ also works on my linux box. Let me know if it breaks any other
+ platform ...
+
+ * res/res_features.c: Fix a typo pointed out by outtolunc, thanks
+ :)
+
+ * channels/chan_sip.c: - Convert initialization of a struct to C99
+ style instead of GNU style - Fix a minor spelling error in a
+ comment
+
+ * res/res_features.c, CHANGES: Update the ParkedCall application to
+ grab the first available parked call if no parked extension is
+ provided as an argument. (closes issue #10803) Reported by:
+ outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by
+ outtolunc (license 237) - modified by me to work a bit
+ differently ...
+
+2007-11-13 19:48 +0000 [r89249] Jason Parker <jparker@digium.com>
+
+ * /, res/res_features.c: Merged revisions 89248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600
+ (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064.
+ It is documented behavior that if a parking extension already
+ exists while using PARKINGEXTEN, dialplan execution will
+ continue. If blind transferring to a Park with PARKINGEXTEN, you
+ must keep this in mind, and handle the failure yourself. Issue
+ 11237, reported by jon. ........
+
+2007-11-13 17:41 +0000 [r89247] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007)
+ | 2 lines If we set a value for qualify, we should actually pay
+ attention to it, instead of overriding the value ........
+
+2007-11-13 16:03 +0000 [r89242] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13
+ Nov 2007) | 5 lines Reverting commit made in revision 89205 since
+ it is unnecessary. Thanks to Kevin for pointing this out ........
+
+2007-11-13 14:03 +0000 [r89240] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/utils.c: Merged revisions 89239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007)
+ | 4 lines Debugging is running into the 16-lock limit. Increase
+ to avoid. (This define is only effective when debugging is turned
+ on, so there's no effect for most installations.) ........
+
+2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_mixmonitor.c: There is the potential to copy
+ uninitialized memory into the mixmonitor->post_process string.
+ This fix prevents that.
+
+ * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12
+ Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only
+ 1 argument is given, then the args.options and args.post_process
+ strings are uninitialized and could contain garbage. This change
+ handles this situation properly by only using arguments that we
+ have parsed. ........
+
+2007-11-13 00:19 +0000 [r89202-89203] Jason Parker <jparker@digium.com>
+
+ * Makefile: oops, somebody left out the directory here...
+
+ * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c,
+ include/asterisk/config.h, res/res_convert.c, res/res_crypto.c,
+ pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h,
+ res/res_config_sqlite.c, res/res_monitor.c,
+ include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen
+ fixes. Also fix a common typo I kept seeing (arguement) in
+ various files. Closes issue #11222, patch by snuffy (with
+ arguement > argument by me).
+
+2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy <murf@digium.com>
+
+ * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake
+ of the mtx_prof stuff.
+
+ * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen
+ update to hashtab.h; closes issue #11223
+
+ * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work,
+ who brought up that these fixes might need to be made.
+
+2007-11-12 20:48 +0000 [r89195] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c, /: Merged revisions 89194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1
+ line Fix a typo pointed out by De_Mon on #asterisk-dev ........
+
+2007-11-12 20:29 +0000 [r89193] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 89191 via svnmerge ........ r89191 |
+ tilghman | 2007-11-12 14:16:18 -0600 (Mon, 12 Nov 2007) | 5 lines
+ If two config writes collide, file corruption could result. Use a
+ mkstemp() file, instead. Reported by: paravoid Patch by: tilghman
+ Closes issue #10781 ........
+
+2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported
+ by: eliel Patches: utils.Makefile.patch uploaded by eliel
+ (modified by me) (license 64)
+
+2007-11-12 18:44 +0000 [r89186] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c,
+ apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in
+ asterisk-dev by Brian Capouch, I determined I too agressive in
+ not initializing arrays passed to pbx_substitute_variables_xxxx;
+ I reviewed the code (again) and hopefully found every possible
+ spot where substitute_variables is called conditionally, and made
+ sure the char array involved was set to a null string.
+
+2007-11-12 17:44 +0000 [r89185] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007)
+ | 5 lines Fix two cases of memory corruption caused by background
+ threads. Reported by: atis Patch by: tilghman Fixes issue #10923
+ ........
+
+2007-11-12 13:36 +0000 [r89178-89179] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged
+ revisions 89173 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) |
+ 1 line if we're NT and no number was dialed and overlapdial is
+ set, we wait for the ISDN timeout instead of starting our own
+ timer. added a comment for the misdn.conf.sample for the
+ overlapdial config option. ........
+
+ * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ Merged revisions 89172 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) |
+ 1 line added restart all interfaces Restart_Indicator, to
+ automatically send a RESTART after the L2 of a PTP Port comes up.
+ Also fixed some places where we have send a RELEASE without need
+ for it. ........
+
+2007-11-12 13:26 +0000 [r89177] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_unistim.c, utils/hashtest.c: Fix building on
+ FreeBSD by including/not including some headers. (closes issue
+ #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys
+ (license 281)
+
+2007-11-12 13:22 +0000 [r89174-89176] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
+ revisions 89171 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) |
+ 1 line fixed a state/event issue with overlapdial=yes when no
+ extension matched. removed the general sending of a
+ RELEASE_COMPLETE when we receive a RELEASE, this is done by
+ mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with
+ mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to
+ upgrade to at least mISDNuser-1.1.6 (when using the NT mode at
+ all) ........
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12
+ Nov 2007) | 1 line fixed the support for CW and therefore for the
+ reject_cause option. ........
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
+ channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
+ revisions 89169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) |
+ 1 line aded ntkeepcalls option, to avoid droÃpping calls when the
+ L2 goes down on a PTP link. There are some pbx which do turn off
+ the L1 for a very short while and restart it immediately.
+ normally T310 should be started and after 10 seconds or so the
+ calls should be dropped, this is a simple fix wihtout this timer.
+ ........
+
+2007-11-09 18:57 +0000 [r89130-89132] Jason Parker <jparker@digium.com>
+
+ * configs/usbradio.conf.sample (added): Add usbradio.conf.sample
+ from branches/1.4/configs - r84162. It was mistakenly deleted in
+ 1.4 without ever being merged to trunk. Reported by eliel on
+ #asterisk-dev.
+
+ * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf
+ (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a
+ few potential deadlocks in cdr_sqlite3_custom. (also rename
+ sample config to .sample) Closes issue #11208, patch by Laureano.
+
+2007-11-09 16:00 +0000 [r89129] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c
+ (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c,
+ include/asterisk/hashtab.h (added), main/config.c: This is the
+ perhaps the biggest, boldest, most daring change I've ever
+ committed to trunk. Forgive me in advance any disruption this may
+ cause, and please, report any problems via the bugtracker. The
+ upside is that this can speed up large dialplans by 20 times (or
+ more). Context, extension, and priority matching are all fairly
+ constant-time searches. I introduce here my hashtables
+ (hashtabs), and a regression for them. I would have used the
+ ast_obj2 tables, but mine are resizeable, and don't need the
+ object destruction capability. The hashtab stuff is well tested
+ and stable. I introduce a data structure, a trie, for extension
+ pattern matching, in which knowledge of all patterns is
+ accumulated, and all matches can be found via a single traversal
+ of the tree. This is per-context. The trie is formed on the first
+ lookup attempt, and stored in the context for future lookups.
+ Destruction routines are in place for hashtabs and the pattern
+ match trie. You can see the contents of the pattern match trie by
+ using the 'dialplan show' cli command when 'core set debug' has
+ been done to put it in debug mode. The pattern tree traversal
+ only traverses those parts of the tree that are interesting. It
+ uses a scoreboard sort of approach to find the best match. The
+ speed of the traversal is more a function of the length of the
+ pattern than the number of patterns in the tree. The tree also
+ contains the CID matching patterns. See the source code comments
+ for details on how everything works. I believe the approach
+ general enough that any issues that might come up involving fine
+ points in the pattern matching algorithm, can be solved by just
+ tweaking things. We shall see. The current pattern matcher is
+ fairly involved, and replicating every nuance of it is difficult.
+ If you find and report problems, I will try to resolve than as
+ quickly as I can. The trie and hashtabs are added to the existing
+ context and exten structs, and none of the old machinery has been
+ removed for the sake of the multitude of functions that use them.
+ In the future, we can (maybe) weed out the linked lists and save
+ some space.
+
+2007-11-08 23:53 +0000 [r89124-89126] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 89125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600
+ (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here..
+ Issue 11203, fix described by vma. ........
+
+ * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be
+ used to check whether it is time to hangup a channel. Closes
+ issue #11202, patch by mnicholson
+
+2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: app_voicemail failed to build when
+ compiling with IMAP_STORAGE Now it does not.
+
+ * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one
+ argument. Thanks to snuffy for pointing this out on IRC
+
+2007-11-08 21:27 +0000 [r89121] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_env.c: Make func_env build again.
+
+2007-11-08 21:01 +0000 [r89120] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov
+ 2007) | 7 lines Rework of the commit I made yesterday to use the
+ already built-in ast_uri_decode function as opposed to my
+ home-rolled one. Also added comments. Thanks to oej for pointing
+ me in the right direction ........
+
+2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_features.c: convert this code to a more efficient
+ idiom
+
+2007-11-08 18:49 +0000 [r89116-89117] Jason Parker <jparker@digium.com>
+
+ * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch
+ by eliel
+
+ * /, configs/cdr_adaptive_odbc.conf.sample,
+ configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600
+ (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using
+ sample configuration files. Issue 11195, patch by eliel. ........
+
+2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan
+ function and deprecate ReadFile.
+
+ * channels/chan_features.c: Fix missed conversion to linkedlists
+ macro change
+
+2007-11-08 16:51 +0000 [r89112] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocking changes from previous 1.4 commit
+
+2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo <rizzo@icir.org>
+
+ * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored
+ anyways).
+
+ * main/audiohook.c: use %d and cast to int instead of %zd for
+ size_t object, this helps portability.
+
+ * channels/chan_unistim.c: initialize a variable to silence
+ compiler. The type of warnings emitted depends on the
+ optimization level, at the lower levels the compiler doesn't
+ always understand what the programmer has in mind. In this case I
+ could not understand it either.
+
+2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/srv.c, /: Merged revisions 89105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007)
+ | 2 lines fix a glaring bug in the new SRV record handling that
+ would cause incorrect weight sorting ........
+
+ * main/autoservice.c, main/frame.c, apps/app_meetme.c,
+ res/res_features.c, funcs/func_strings.c, main/devicestate.c,
+ res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
+ codecs/codec_zap.c, res/res_jabber.c, main/indications.c,
+ main/astobj2.c, main/config.c, main/loader.c, main/cli.c,
+ main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c,
+ main/logger.c, main/app.c, main/image.c, res/res_speech.c,
+ main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c,
+ channels/chan_agent.c, utils/astman.c, apps/app_queue.c,
+ channels/iax2-parser.c, main/srv.c,
+ include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c,
+ main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c,
+ apps/app_voicemail.c: improve linked-list macros in two ways: -
+ the *_CURRENT macros no longer need the list head pointer
+ argument - add AST_LIST_MOVE_CURRENT to encapsulate the
+ remove/add operation when moving entries between lists
+
+2007-11-08 05:00 +0000 [r89104] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007)
+ | 2 lines Typo ........
+
+2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4
+ lines Do not add a sip: to the beginning of the To URI unless
+ needed. (closes issue #10756) Reported by: goestelecom ........
+
+ * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6
+ lines Improve the devicestate logic for multiple devices. If any
+ are available then the extension is considered available. (closes
+ issue #10164) Reported by: nic_bellamy Patches:
+ sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8
+ lines Add support for allowing one outgoing transaction. This
+ means if a response comes back out of order chan_sip will still
+ handle it. I dream of a chan_sip with real transaction support.
+ (closes issue #10946) Reported by: flefoll (closes issue #10915)
+ Reported by: ramonpeek (closes issue #9567) Reported by:
+ atca_pres ........
+
+ * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4
+ lines If callerid is configured in sip.conf use that for checking
+ the presence of an extension in the dialplan. (closes issue
+ #11185) Reported by: spditner ........
+
+2007-11-07 23:47 +0000 [r89094] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007)
+ | 7 lines The member refcount must be incremented, to avoid using
+ it after deallocation. A huge thanks go to lvl- for patiently
+ providing the necessary valgrind output that was necessary to
+ finding this problem of memory corruption. Reported by: lvl-
+ Patch by: tilghman Closes issue #11174 ........
+
+2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: If imapfolder has been specified in
+ voicemail.conf, we should not connect to INBOX... ever. It may
+ not exist. (closes issue #11151, reported by selsky, patched by
+ me)
+
+ * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov
+ 2007) | 6 lines This patch makes it possible for SIP phones to
+ dial extensions defined with '#' characters in extensions.conf
+ AND maintain their escaped characters when forming URI's (closes
+ issue #10681, reported by cahen, patched by me, code review by
+ file) ........
+
+2007-11-07 22:09 +0000 [r89089] Steve Murphy <murf@digium.com>
+
+ * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1
+ line In response to 10578, I just ran 1.4 thru valgrind; some of
+ the config leakage I've already fixed, but it doesn't hurt to
+ double check. I found and fixed leaks in res_jabber, cdr_tds,
+ pbx_ael. Nothing major, tho. ........
+
+2007-11-07 17:45 +0000 [r89086] Joshua Colp <jcolp@digium.com>
+
+ * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds
+ again.
+
+2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: remove enter/exit comments when handling subdirectory.
+ If we really want them we can remove the --no-print-directory
+
+ * main/loader.c: remove a debugging message which i forgot in.
+
+ * Makefile: match changes in menuselect's Makefile
+
+2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c: Suppress erroneous warnings on load.
+ Reported by: eliel Patch by: eliel Closes issue #11177
+
+ * /, configs/extensions.ael.sample: Merged revisions 89079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007)
+ | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch
+ by: eliel Closes issue #11178 ........
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: Provide the
+ ability to directly manipulate the TON/NPI bits in the
+ dialstring. Reported by: thetatag Patch by:
+ thetatag/stevens/tilghman Closes issue #5331
+
+ * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy,
+ which provides FastAGI functionality for EAGI, while also
+ buffering the audio stream. Reported by: devil_slayer Patch by:
+ devil_slayer Closes issue #8921
+
+2007-11-07 00:16 +0000 [r89076] Russell Bryant <russell@digium.com>
+
+ * main/astmm.c: Fix another CLI command so it doesn't run the real
+ code when called for initialization.
+
+2007-11-07 00:04 +0000 [r89075] Mark Michelson <mmichelson@digium.com>
+
+ * doc/tex/imapstorage.tex: Adding documentation regarding
+ imapfolder, imapgreetings, and greetingsfolder options in
+ voicemail.conf (closes issue #11133, reported by selsky, patched
+ by blitzrage)
+
+2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the
+ channel name as a prefix to the "agi debug" output. This makes
+ AGI debugging on busy systems much easier. (closes issue #10730)
+ Reported by: junky Patches: agi_debug_chan.diff uploaded by junky
+ (license 177) 20070923_10730.diff uploaded by mvanbaak (license
+ 7)
+
+ * apps/app_meetme.c, CHANGES: Added the ability to do "meetme
+ concise" with the "meetme" CLI command. This extends the concise
+ capabilities of this CLI command to include listing all
+ conferences, instead of an addition to the other sub commands for
+ the "meetme" command. (closes issue #11078) Reported by: jthomas
+ Patches: meetme-concise.patch uploaded by jthomas (license 293)
+
+2007-11-06 23:08 +0000 [r89072] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Fix up some PBX logic that became broken. The code
+ would exit prematurely when it should have been collecting more
+ digits. (closes issue #11175) Reported by: pj
+
+2007-11-06 22:51 +0000 [r89071] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_jingle.c, channels/chan_phone.c,
+ codecs/codec_g722.c, main/frame.c, channels/chan_sip.c,
+ channels/chan_skinny.c, main/translate.c, channels/chan_h323.c,
+ main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h,
+ main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h:
+ Commit some cleanups to the format type code. - Remove the
+ AST_FORMAT_MAX_* types, as these are consuming 3 out of our
+ available 32 bits. - Add a native slin16 type, so that 16kHz
+ codecs can translate without losing resolution. (This doesn't
+ affect anything immediately, until another codec has wb support.)
+
+2007-11-06 22:36 +0000 [r89070] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
+ queue strategy wrandom (closes issue #10942, reported and patched
+ by julianjm, documentation changes by me)
+
+2007-11-06 22:15 +0000 [r89069] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added
+ the S() and L() options to the MeetMe application. These are
+ pretty much identical to the S() and L() options to Dial(). They
+ let you set timeouts for the conference, as well as have warning
+ sounds played to let the caller know how much time is left, and
+ when it is running out. (closes issue #8030) Reported by: areski
+ Patches: meetme_timeout_timelimit_v2.patch uploaded by areski
+ (license 29)
+
+2007-11-06 22:05 +0000 [r89068] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Added CLI and manager commands for changing a
+ queue member's penalty (closes issue #9374, reported and
+ initially patched by wuwu, intermediate patch by eliel, and final
+ patch by me)
+
+2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add some more locking as well as API update
+ for libss7 for new transport types
+
+2007-11-06 21:08 +0000 [r89062] Steve Murphy <murf@digium.com>
+
+ * /, main/config.c: Merged revisions 89036 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1
+ line closes issue #8786 - where the [catname](!) and
+ [catname](othercat1,othercat2,...) notation gets dropped across a
+ ConfigUpdate (or any other thing that would cause a config file
+ to be written). While I was at it, I also cleaned up some of the
+ destroy routines to free up comments, which was not being done.
+ Made sure the new struct I introduced is also cleaned up properly
+ at destruction time. My code handles multiple template
+ inclusions. Many thanks to ssokol for his patch, which, while not
+ literally used in the final merge, served as a foundation for the
+ fix. ........
+
+2007-11-06 20:55 +0000 [r89057] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Remove native bridging check for DTMF based
+ transfers. Thanks to the last batch of RTP changes it is no
+ longer required for the media stream to go through Asterisk if
+ DTMF is going over signalling. It will simply reinvite back as
+ needed. (closes issue #11172) Reported by: ibc
+
+2007-11-06 20:32 +0000 [r89055] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_features.c: Instead of trying to callback a local channel
+ on a failed attended transfer, call the device that made the
+ transfer instead. This makes for much smoother calling back when
+ queues are involved. (closes issue #11155, reported by IPetrov)
+ Tremendous thanks to Russell for pulling me out of my block I was
+ having on this one
+
+2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06
+ Nov 2007) | 3 lines Fix init_classes() so that classes that
+ actually do have files loaded aren't treated as empty, and
+ immediately destroyed ... ........
+
+ * main/astmm.c: Fix the memory show allocations CLI command so that
+ it doesn't spew out all of the current memory allocations when
+ you start Asterisk, when the command's handler gets called for
+ initialization.
+
+2007-11-06 19:40 +0000 [r89051] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in
+ OSX for a problem blitzrage found
+
+2007-11-06 19:23 +0000 [r89050] Olle Johansson <oej@edvina.net>
+
+ * main/fskmodem.c: Formatting. Illegaly using some spare spaces
+ from Russell's space-bucket.
+
+2007-11-06 19:16 +0000 [r89049] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06
+ Nov 2007) | 2 lines We went to the trouble of creating a method
+ of tracking failed trylocks, then never turned it on (oops).
+ ........
+
+2007-11-06 19:10 +0000 [r89048] Olle Johansson <oej@edvina.net>
+
+ * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes
+ (preparing for SIP changes - adding TDD support to SIP)
+
+2007-11-06 19:10 +0000 [r89047] Jason Parker <jparker@digium.com>
+
+ * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4
+ lines Correctly set the total number of channels from a zaptel
+ transcoder board. SPD-49, patch by Matthew Nicholson. ........
+
+2007-11-06 19:04 +0000 [r89044] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c,
+ apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c,
+ apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c,
+ apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c,
+ apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c,
+ apps/app_softhangup.c, apps/app_waitforsilence.c,
+ channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c,
+ apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c,
+ apps/app_forkcdr.c, apps/app_waitforring.c,
+ apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c,
+ res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+ apps/app_stack.c: "show application <foo>" changes for clarity.
+ (closes issue #11171, reported and patched by blitzrage) Many
+ thanks!
+
+2007-11-06 19:04 +0000 [r89043] Olle Johansson <oej@edvina.net>
+
+ * /, main/tdd.c: Merged revisions 89042 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2
+ lines Bug fixes to tdd support in zaptel. ........ (Small changes
+ for trunk)
+
+2007-11-06 18:44 +0000 [r89041] Jason Parker <jparker@digium.com>
+
+ * channels/chan_jingle.c, include/asterisk/jabber.h,
+ channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle
+ to use TLS connections again. Closes issue #9972
+
+2007-11-06 18:23 +0000 [r89038] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06
+ Nov 2007) | 11 lines If someone were to delete the files used by
+ an existing MOH class, and then issue a reload, further use of
+ that class could result in a crash due to dividing by zero. This
+ set of changes fixes up some places to prevent this from
+ happening. (closes issue #10948) Reported by: jcomellas Patches:
+ res_musiconhold_division_by_zero.patch uploaded by jcomellas
+ (license 282) Additional changes added by me. ........
+
+2007-11-06 17:10 +0000 [r89034] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4
+ lines Make it so that if a peer is determined to be unreachable
+ using qualify their devicestate will report back unavailable.
+ (closes issue #11006) Reported by: pj ........
+
+2007-11-06 17:05 +0000 [r89031] Luigi Rizzo <rizzo@icir.org>
+
+ * main/loader.c: Fix embedding of modules on FreeBSD: the
+ constructor for the list of modules was run after the
+ constructors for the embedded modules (which appended entries to
+ the list). As a result, the list appeared empty when it was time
+ to use it. On linux the order of execution of constructor was
+ evidently different (it may depend on the ordering of modules in
+ the ELF file). This is only a workaround - there may be other
+ situations where the execution of constructors causes problems,
+ so if we manage to find a more general solution this workaround
+ can go away.
+
+2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged
+ revisions 88994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6
+ lines Fix improbable but possible memory leaks in chan_zap.
+ (closes issue #11166) Reported by: eliel Patches:
+ chan_zap.c.patch uploaded by eliel (license 64) ........
+
+ * channels/chan_agent.c: Update chan_agent documentation. Change a
+ | to , as that is now the required way. (closes issue #11167)
+ Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel
+ (license 64)
+
+2007-11-06 15:01 +0000 [r88973] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_unistim.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Set up detection
+ of IP_PKTINFO in autoconf for chan_unistim
+
+2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c: convert uses of LOG_DEBUG to use
+ ast_debug()
+
+ * channels/chan_unistim.c, configs/unistim.conf.sample: Add
+ jitterbuffer support to chan_unistim. (closes issue #11168)
+ Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by
+ IgorG (license 20)
+
+ * main/pbx.c, /, channels/busy.h, channels/ringtone.h,
+ include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) |
+ 12 lines After seeing crashes related to channel variables, I
+ went looking around at the ways that channel variables are
+ handled. In general, they were not handled in a thread-safe way.
+ The channel _must_ be locked when reading or writing from/to the
+ channel variable list. What I have done to improve this situation
+ is to make pbx_builtin_setvar_helper() and friends lock the
+ channel when doing their thing. Asterisk API calls almost all
+ lock the channel for you as necessary, but this family of
+ functions did not. (closes issue #10923, reported by atis)
+ (closes issue #11159, reported by 850t) ........
+
+ * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06
+ Nov 2007) | 8 lines Remove some checks to see if locks are
+ initialized from the non-DEBUG_THREADS versions of the lock
+ routines. These are incorrect for a number of reasons: - It
+ breaks the build on mac. - If there is a problem with locks not
+ getting initialized, then the proper fix is to find that place
+ and fix the code so that it does get initialized. - If additional
+ debug code is needed to help find the problem areas, then this
+ type of things should _only_ be put in the DEBUG_THREADS
+ wrappers. ........
+
+2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/Makefile: explain that the host environment must be used
+ to build gentone; Remove unset variables, they would be
+ misleading.
+
+ * Makefile: don't export variables that can be retrieved from
+ makeopts in child subdirs
+
+2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05
+ Nov 2007) | 2 lines update comment to match the state of the code
+ ........
+
+2007-11-05 23:31 +0000 [r88827] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 88826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov
+ 2007) | 6 lines Reworked deadlock avoidance in __ast_read.
+ Restored audio to callback agents. (closes issue #11071, reported
+ by callguy, patched by me, tested by callguy and Ted Brown)
+ ........
+
+2007-11-05 21:36 +0000 [r88770] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level
+ Makefile. This too is used only in one place.
+
+2007-11-05 21:35 +0000 [r88769] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) |
+ 8 lines When traversing the list of channel variables here in
+ transmit_invite(), the asterisk channel must be locked, as this
+ data may change at any time. (I have seen numerous reports of
+ crashes related to the handling of channel variables. There are a
+ couple of issues on the bug tracker related to it, but it has
+ also been noted on IRC and mailing lists. So, I am finding and
+ fixing some places where channel variables are handled
+ improperly.) ........
+
+2007-11-05 21:27 +0000 [r88767] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile, main/Makefile: Move the last instance of AST_LIBS to
+ the only place it is used, namely main/Makefile . I am unclear
+ where decisions on the build environment (CFLAGS, LDFLAGS, LIBS
+ and so on) should be made - right now they are split here and
+ there. As a first step in cleaning up this situation, i am trying
+ to at least collect all instances of each variable in one place.
+
+2007-11-05 21:23 +0000 [r88766] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) |
+ 2 lines Fix up some indentation. ........
+
+2007-11-05 20:50 +0000 [r88764] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile.moddir_rules: comment out an unused variable. Remove it
+ in a few days if no problems arise.
+
+2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant <russell@digium.com>
+
+ * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) |
+ 7 lines Merge changes from
+ asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
+ record support in Asterisk was broken. There was no guarantee on
+ what record Asterisk would choose to actually use. This set of
+ changes improves the situation by ensuring that Asterisk will
+ choose the highest priority record. ........
+
+ * main/channel.c, /: Merged revisions 88709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) |
+ 20 lines Merge the last bit of changes from
+ asterisk/team/russell/readq-1.4 The issue here is that the
+ channel frame readq handling got broken when the code was
+ converted to use the linked list macros. It caused corruption of
+ the list head and tail pointers. So, I fixed up the usage of the
+ linked list macros and in passing, simplified the code. I also
+ documented what the code is doing, as it was a bit difficult to
+ figure out at first. This bug showed itself with crashes showing
+ messed up head/tail pointers for the readq. However, there are a
+ couple of crashes that aren't quite as obvious, but I think may
+ be related. So, if your bug gets closed by this commit, but you
+ still have a problem, please reopen or create a new bug report.
+ (closes issue #10936) (closes issue #10595) (closes issue #10368)
+ (closes issue #11084) (closes issue #10040) (closes issue #10840)
+ ........
+
+2007-11-05 19:22 +0000 [r88675] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: Cleanup the installation of samples, avoiding
+ repetitions. I am preserving the behaviour on *.adsi files, i.e.
+ overwrite anything there without making a backup. However I am
+ not sure that this is the intended behaviour.
+
+2007-11-05 18:52 +0000 [r88673] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7
+ lines If a SIP channel is put on hold multiple times do not keep
+ incrementing the onHold value. (closes issue #11085) Reported by:
+ francesco_r Tested by: blitzrage (closes issue #10474) Reported
+ by: acennami ........
+
+2007-11-05 18:22 +0000 [r88653] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: Change wording to that suggested by MasterYoda
+
+2007-11-05 18:00 +0000 [r88652] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned
+ output.
+
+2007-11-05 17:52 +0000 [r88651] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 88624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) |
+ 5 lines Fix up datastore handling in ast_do_masquerade(). The
+ code is intended to move any channel datastores from the old
+ channel to the new one. However, it did not use the linked list
+ macros properly to accomplish the task. The existing code would
+ only work if there was only a single datastore on the old
+ channel. ........
+
+2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile: print messages when entering/leaving a directory so we
+ know where we are (sometimes it is obvious, sometimes it is not).
+
+ * Makefile.moddir_rules: merge two rules with the same right hand;
+ document a bit what is done here.
+
+2007-11-05 17:21 +0000 [r88586] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600
+ (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config
+ structure on configuration failure. Issue 11163, patch by eliel.
+ ........
+
+2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile.rules: use a variable name that actually indicates what
+ it is for
+
+2007-11-05 16:41 +0000 [r88553] Luigi Rizzo <rizzo@icir.org>
+
+ * Makefile.rules: Put extra compiler flags into a variable so they
+ are not repeated too many times. On passing, add some comments
+ and fix indentation a bit. On passing, i suspect that the
+ following pattern is wrong %.eoo: %.o but in case it will be
+ fixed in a later commit.
+
+2007-11-05 16:30 +0000 [r88540] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007)
+ | 4 lines Don't check used pooled connections for connection
+ status, as it will cause issues for prepared queries. Reported
+ by: Nick Gorham (via -dev list) Patch by: tilghman ........
+
+2007-11-05 15:15 +0000 [r88525] Luigi Rizzo <rizzo@icir.org>
+
+ * main/db.c: remove a cygwin-specific function remap that does not
+ work.
+
+2007-11-05 13:11 +0000 [r88510] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_unistim.c: Fix memory leaks and deadlocks in
+ chan_unistim. (closes issue #11158) Reported by: eliel Patches:
+ chan_unistim.c.patch uploaded by eliel (license 64)
+
+2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo <rizzo@icir.org>
+
+ * /: block merging of not-applicable patch
+
+ * main/channel.c, main/pbx.c, apps/app_meetme.c,
+ channels/chan_sip.c, res/res_features.c, main/utils.c,
+ channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify
+ the implementation and the API for stringfields; details and
+ examples are in include/asterisk/stringfields.h. Not applicable
+ to older branches except for 1.4 which will receive a fix for the
+ routines that free memory pools.
+
+2007-11-03 14:19 +0000 [r88437] Tilghman Lesher <tlesher@digium.com>
+
+ * main/term.c: Revert commit #86119. Some users intentionally do
+ not want colorized terminals, so this was a misfeature.
+
+2007-11-03 04:55 +0000 [r88422] James Golovich <james@gnuinter.net>
+
+ * main/db.c: Set CLI command to the correct name. Rev 85460
+ introduced two 'database show' commands when this one should have
+ been 'database showkey'
+
+2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c: fix some issues with crashing on unload,
+ when it didn't completely load cleanly
+
+ * channels/chan_unistim.c: Convert the CLI commands to the new
+ format
+
+ * pbx/pbx_lua.c: propagate the DECLINE return value back to the
+ loader
+
+ * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not
+ present.
+
+ * main/cli.c: Show the channel unique ID in the "show channel
+ concise" output (closes issue #11148, requested by falves11,
+ patched by me)
+
+ * channels/chan_unistim.c (added), CREDITS,
+ configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt
+ (added): Merge the code from asterisk/team/group/chan_unistim:
+ This introduces a new channel driver, chan_unistim, that supports
+ the Unistim VoIP protocol for Nortel phones. The following models
+ have been confirmed to work: i2002, i2004 and i2050. (closes
+ issue #8864) Reported by: c_hans Patches: chan_unistim.patch
+ uploaded by c (license 304) ustm_no_conf.diff uploaded by junky
+ (license 177) Tested by: c_hans, dbowerman, math, junky, loloski
+
+2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4
+ lines Make subscribecontext behave as advertised. It will now
+ look for the presence of a hint in the given context (be it
+ subscribecontext or context). (closes issue #10702) Reported by:
+ slavon ........
+
+ * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6
+ lines If an INFO request within a dialog is received with a
+ content length of 0 simply send back a 200 OK. It is valid to do
+ this and the remote side is probably using it to make sure the
+ signalling is still alive. (closes issue #5747) Reported by:
+ chandi Patches: infofix-81430-1.patch uploaded by IgorG (license
+ 20) ........
+
+2007-11-02 20:13 +0000 [r88327] Russell Bryant <russell@digium.com>
+
+ * doc/tex/Makefile: Fix replacing the version number when it has a
+ '/' in it, like SVN-group-chan_unistim-r88326M-/trunk
+
+2007-11-02 17:34 +0000 [r88287] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90
+
+2007-11-02 16:54 +0000 [r88284] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 88283 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500
+ (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a
+ language to ast_fileexists, otherwise it may fail for anything
+ besides en Issue 11147, fix discovered by both citats and myself
+ (independently), with input from Corydon76 ........
+
+2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: Add a few bytes on LUA
+
+ * main/pbx.c, utils/build-extensions-conf.lua (added),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c
+ (added), configs/extensions.lua.sample (added),
+ include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of
+ doing extensions Reported by: mnicholson Patch by: mnicholson
+ Closes issue #11140
+
+ * main/config.c: Don't re-cache the filename, but check to see if
+ it already exists Reported by: jamesgolovich Patch by:
+ jamesgolovich Closes issue #11144
+
+ * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02
+ Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy
+ Patch by: ys Closes issue #11143 ........
+
+ * main/pbx.c: 'h' extension doesn't execute past first priority
+ Reported by: dimas Patch by: dimas Closes bug #11146
+
+2007-11-02 03:09 +0000 [r88197] Joshua Colp <jcolp@digium.com>
+
+ * cdr/cdr_odbc.c: Restore building under 64-bit platforms.
+
+2007-11-01 23:26 +0000 [r88184] Jason Parker <jparker@digium.com>
+
+ * channels/chan_jingle.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in:
+ Remove traces of gnutls, since we no longer use/need it.
+
+2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Modify WaitExten to include an optional dialtone
+ Closes issue #10783
+
+ * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc
+ managed connections Closes issue #10614
+
+2007-11-01 22:26 +0000 [r88166] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c,
+ funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c,
+ apps/app_exec.c, apps/app_queue.c, apps/app_playback.c,
+ res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c,
+ apps/app_minivm.c, res/res_agi.c, main/logger.c,
+ pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c,
+ utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c,
+ apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c,
+ cdr/cdr_manager.c: This commits the performance mods that give
+ the priority processing engine in the pbx, a 25-30% speed boost.
+ The two updates used, are, first, to merge the
+ ast_exists_extension() and the ast_spawn_extension() where they
+ are called sequentially in a loop in the code, into a slightly
+ upgraded version of ast_spawn_extension(), with a few extra args;
+ and, second, I modified the substitute_variables_helper_full, so
+ it zeroes out the byte after the evaluated string instead of
+ demanding you pre-zero the buffer; I also went thru the code and
+ removed the code that zeroed this buffer before every call to the
+ substitute_variables_helper_full. The first fix provides about a
+ 9% speedup, and the second the rest. These figures come from the
+ 'PIPS' benchmark I describe in blogs, conf. reports, etc.
+
+2007-11-01 22:19 +0000 [r88164-88165] Jason Parker <jparker@digium.com>
+
+ * /: Crap, accidentally copied the props. Thanks for pointing this
+ out mvanbaak. The odds are quite high that this will break
+ automerge on every team branch.
+
+ * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber
+ to use openssl rather than gnutls. Closes issue #9972, patch by
+ phsultan. Copied from branch at
+ http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/
+
+2007-11-01 17:25 +0000 [r88117] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01
+ Nov 2007) | 2 lines Add some notes on using valgrind ........
+
+2007-11-01 16:22 +0000 [r88079] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4
+ lines Make sure we set the poll fds to NULL after free()ing it.
+ Part of issue 11017, patch by tzafrir. ........
+
+2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free()
+ to ast_free(). (No functional differences.) (closes issue #11138)
+ Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel
+ (license 64) chan_sip.c.patch uploaded by eliel (license 64)
+
+ * utils/Makefile: Remove another copied source file on "make
+ clean". (closes issue #11137) Reported by: IgorG Patches:
+ addonclean-87971-1.patch uploaded by IgorG (license 20)
+
+2007-11-01 13:30 +0000 [r88027] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2
+ lines Fix up commit for my Zap channel with spies in Meetme fix.
+ (thanks Tony Mountifield!) ........
+
+2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher <tlesher@digium.com>
+
+ * main/utils.c: Conditionally free lock_info->thread_name to avoid
+ a useless warning Reported by: snuffy Patch by: snuffy Closes
+ issue #11125
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to
+ pair calls of ast_malloc and ast_calloc Reported by: eliel Patch
+ by: eliel Closes issue #11135
+
+ * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed
+ by: tilghman Closes issue #11136
+
+2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4
+ lines If a Zap channel contains a spy or a spy is added take it
+ out of the conference in kernel space and make it go through
+ Asterisk so the spy gets audio from both sides. (closes issue
+ #10060) Reported by: mparker ........
+
+ * main/pbx.c: Drop any more references to type in the Exception
+ dialplan function. (closes issue #11134) Reported by: blitzrage
+ Patches: exception_patch.txt uploaded by blitzrage (license 10)
+
+2007-10-31 21:23 +0000 [r87889-87909] Jason Parker <jparker@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500
+ (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated
+ memory before returning. Issue 11131, patch by eliel. ........
+
+ * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
+ revisions 87906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11130) (closes issue #11132) ........ r87906 | qwell |
+ 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try
+ to allocate memory that we're just going to re-allocate later
+ anyways. Issues 11130 and 11132, patch by eliel. ........
+
+ * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c,
+ formats/format_wav_gsm.c, res/res_musiconhold.c,
+ codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c,
+ formats/format_pcm.c, formats/format_h263.c,
+ formats/format_h264.c, formats/format_jpeg.c,
+ formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c,
+ codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
+ formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c,
+ res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c,
+ formats/format_ogg_vorbis.c, res/res_ael_share.c,
+ formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c,
+ res/res_indications.c, codecs/codec_ilbc.c,
+ formats/format_g726.c, formats/format_g729.c: More changes to
+ change return values from load_module functions. (issue #11096)
+ Patches: codec_adpcm.c.patch uploaded by moy (license 222)
+ codec_alaw.c.patch uploaded by moy (license 222)
+ codec_a_mu.c.patch uploaded by moy (license 222)
+ codec_g722.c.patch uploaded by moy (license 222)
+ codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch
+ uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy
+ (license 222) codec_lpc10.c.patch uploaded by moy (license 222)
+ codec_speex.c.patch uploaded by moy (license 222)
+ codec_ulaw.c.patch uploaded by moy (license 222)
+ codec_zap.c.patch uploaded by moy (license 222)
+ format_g723.c.patch uploaded by moy (license 222)
+ format_g726.c.patch uploaded by moy (license 222)
+ format_g729.c.patch uploaded by moy (license 222)
+ format_gsm.c.patch uploaded by moy (license 222)
+ format_h263.c.patch uploaded by moy (license 222)
+ format_h264.c.patch uploaded by moy (license 222)
+ format_ilbc.c.patch uploaded by moy (license 222)
+ format_jpeg.c.patch uploaded by moy (license 222)
+ format_ogg_vorbis.c.patch uploaded by moy (license 222)
+ format_pcm.c.patch uploaded by moy (license 222)
+ format_sln.c.patch uploaded by moy (license 222)
+ format_vox.c.patch uploaded by moy (license 222)
+ format_wav.c.patch uploaded by moy (license 222)
+ format_wav_gsm.c.patch uploaded by moy (license 222)
+ res_adsi.c.patch uploaded by eliel (license 64)
+ res_ael_share.c.patch uploaded by eliel (license 64)
+ res_clioriginate.c.patch uploaded by eliel (license 64)
+ res_convert.c.patch uploaded by eliel (license 64)
+ res_indications.c.patch uploaded by eliel (license 64)
+ res_musiconhold.c.patch uploaded by eliel (license 64)
+ res_smdi.c.patch uploaded by eliel (license 64)
+ res_speech.c.patch uploaded by eliel (license 64)
+
+2007-10-31 18:53 +0000 [r87888] Steve Murphy <murf@digium.com>
+
+ * /: Merged revisions 87849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1
+ line closes issue #11108 -- where the 'dialplan save' cli command
+ saves a file where the semicolon is not escaped. Fixed this; User
+ also wanted comments to be preserved across dialplan save, but
+ this is impossible at this point in time, because comments are
+ not stored in the dialplan. They are 'compiled' out of
+ extensions.conf. The only way to preserve those comments is to
+ use the config file reader/writer that the GUI uses to allow
+ online user edits. extensions.conf is first and foremost, a
+ config file, and is read in by the normal config-file reading
+ routines. Then, it is processed into a dialplan (context/exten
+ structs). (in the case of trunk, tho, no mods needed to be made
+ -- works OK there -- just make sure you use ',' to sep app args!)
+ ........
+
+2007-10-31 18:09 +0000 [r87854] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, /: Merged revisions 87852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007)
+ | 2 lines Create samples for ALL of the available options in
+ asterisk.conf ........
+
+2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Add volume adjustment in.
+
+ * apps/app_mixmonitor.c: Restore operation of the option that only
+ writes when the channel is bridged.
+
+ * apps/app_chanspy.c: Add volume adjustment to spy audiohook in
+ app_chanspy.
+
+2007-10-31 16:13 +0000 [r87817] Tilghman Lesher <tlesher@digium.com>
+
+ * CREDITS: Formatting cleanups, remove obsolete contributions
+ (modules no longer in Asterisk), and obfuscate email addresses
+ enough to stop most spam harvesters.
+
+2007-10-31 16:07 +0000 [r87815] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/channel.h: Remove old whisper remnants from
+ channel.h
+
+2007-10-31 15:46 +0000 [r87811] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Optimize pbx_substitute_variables
+
+2007-10-31 04:20 +0000 [r87776] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1
+ line Included some verbage in the check_includes func, to inform
+ the user that included contexts that have no match in the AEL,
+ might be OK, as AEL cannot check in the extensions.conf or the
+ in-memory contexts, as they may not be there at the time of the
+ check. ........
+
+2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30
+ Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD
+ Reported by: ys Fixed by: ys Closes issue #11116 ........
+
+ * apps/app_exec.c: If no '?' is found in the arguments, don't
+ attempt to continue. Reported by: blitzrage Fixed by: tilghman
+ Closes issue #11111
+
+2007-10-30 21:22 +0000 [r87687] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) |
+ 11 lines Merge the changes from team/russell/iax2_poke_fix and
+ iax2-poke-fix-trunk There was a race condition related to the
+ handling of POKEing peers. Essentially, a reference to a peer is
+ held by the scheduler when there are pending callbacks, but the
+ reference count didn't reflect it. So, it was possible for a peer
+ to hit a reference count of zero and have its destructor begin to
+ be called at the same time that the scheduler thread ran a POKE
+ related callback. If that happened, a crash would likely occur.
+ (closes issue #11082, closes issue #11094) ........
+
+2007-10-30 20:30 +0000 [r87626-87651] Jason Parker <jparker@digium.com>
+
+ * /, channels/Makefile: Merged revisions 87650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1
+ line Only try to clean out h323/ if the h323/Makefile exists.
+ ........
+
+ * main/pbx.c: Update documentation to give an example of how to use
+ the return status of RaiseException Closes issue #11117, patch by
+ blitzrage (yay blitzrage)
+
+2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: The priority gets incremented after raising an
+ exception, so the priority should be set to 0
+
+ * main/pbx.c: Jumped the gun a bit in the RaiseException app. It
+ would always return -1 since it checked for the existence of
+ something that will never exist.
+
+2007-10-30 16:15 +0000 [r87572] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_features.c: Merged revisions 87571 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4
+ lines Add two more checks before printing out a warning message
+ about bridging. If either channel has hungup of course the bridge
+ will have failed. (closes issue #10009) Reported by: dimas
+ ........
+
+2007-10-30 15:47 +0000 [r87568] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500
+ (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris.
+ Issue 11113, patch by snuffy. ........
+
+2007-10-30 15:11 +0000 [r87535] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 87534 via svnmerge ........ r87534 | file |
+ 2007-10-30 12:10:13 -0300 (Tue, 30 Oct 2007) | 2 lines Return 1.4
+ to a state where it builds. Changing the arguments to a function
+ and not changing where they are used is bad, mmmk? ........
+
+2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to
+ fix #10995, I found all kinds of cruft in this Makefile. It
+ should all be gone now, and as a side effect hashtest2 now builds
+ with --enable-dev-mode enabled without a host of errors
+
+ * agi/Makefile, utils/Makefile, codecs/g722/Makefile,
+ main/editline/Makefile.in, Makefile.moddir_rules,
+ codecs/ilbc/Makefile, codecs/lpc10/Makefile,
+ main/db1-ast/Makefile: clean up assembler and preprocessor files
+ if they are here too
+
+ * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs,
+ codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main,
+ codecs/gsm, main/minimime, pbx, res, channels: ignore
+ preprocessor and assembler files if they are present
+
+ * Makefile, /: Merged revisions 87460 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007)
+ | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is
+ already there (used when debugging preprocessor issues) because
+ the compiler will whine about each compile command ........
+
+2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: If a caller is listen-only, then don't bother
+ with doing talker detection. (closes issue #10911, reported by
+ junky, patched by me)
+
+ * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) |
+ 5 lines Add some more details to the output of "core show locks".
+ When a thread is waiting for a lock, this will now show the
+ details about who currently has it locked. (inspired by issue
+ #11100) ........
+
+2007-10-29 20:13 +0000 [r87395] Mark Michelson <mmichelson@digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c: Adding the more flexible
+ QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function.
+ A deprecation notice will be issued the first time
+ QUEUE_MEMBER_COUNT is used.
+
+2007-10-29 20:02 +0000 [r87394] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Drop the RTCP Read too short message to debug. There
+ are some phones out there that send a sort of keep alive packet
+ in the RTCP that trigger this every 5 seconds.
+
+2007-10-29 19:56 +0000 [r87393] Jason Parker <jparker@digium.com>
+
+ * apps/app_record.c: Make sure we set flags to a 0 value before
+ trying to use it. Pointed out by seanbright while I was debugging
+ issue 11109.
+
+2007-10-29 19:47 +0000 [r87392] Russell Bryant <russell@digium.com>
+
+ * /, main/astmm.c: Merged revisions 87373 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) |
+ 5 lines Remove a lock that doesn't make any sense. The regions
+ lock needs to be held when traversing the list of allocated
+ chunks so that they can be printed out to the CLI. (Thanks to
+ eliel on #asterisk-dev for pointing this out!) ........
+
+2007-10-29 17:22 +0000 [r87343] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6
+ lines Fix issue where if both sides of the dialog cancelled the
+ dialog at the same time chan_sip could kepe retransmitting a
+ response for no reason. (closes issue #9566) Reported by:
+ atca_pres Patches: bug9566.patch uploaded by oej ........
+
+2007-10-29 17:19 +0000 [r87341] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 87340 via svnmerge (Closes issue #11104)
+ ........ r87340 | qwell | 2007-10-29 12:13:04 -0500 (Mon, 29 Oct
+ 2007) | 4 lines Allow some function modules to compile under dev
+ mode. Issue 11104, patch by andrew. ........
+
+2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes
+ issue #11105) Reported by: eliel Patches: app_voicemail.c.patch
+ uploaded by eliel (license 64)
+
+ * channels/chan_misdn.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Add autoconf
+ checks for extra suppserv definitions that are not present in
+ releases yet. chan_misdn should now build against the latest
+ release. (closes issue #11103) Reported by: IgorG
+
+ * /, main/utils.c: Merged revisions 87294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6
+ lines Fix issue with ast_unescape_semicolon going into an endless
+ loop. (closes issue #10550) Reported by: ramonpeek Patches:
+ unescape-85177-1.patch uploaded by IgorG (license 20) ........
+
+2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_dialgroup.c (added): Add a simple dialgroup function.
+ By taking one of the simpler uses of Queue away from Queue, we
+ simplify the lives of people who do not need all the bells and
+ whistles. Also, this is part of the functions that people need to
+ reimplement Queue in the dialplan, as a set of logic, rather than
+ as a single app with hundreds of options.
+
+ * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c,
+ funcs/func_realtime.c: Merged revisions 87262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007)
+ | 7 lines Add autoservice to several more functions which might
+ delay in their responses. Also, make sure that func_odbc
+ functions have a channel on which to set variables. Reported by
+ russell Fixed by tilghman Closes issue #11099 ........
+
+2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Update the configure script for the last
+ libss7 API change
+
+ * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists
+ before attempting to start or stop channel autoservice in
+ func_lock and func_shell.
+
+2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add Circuit Group Queury message code
+
+ * channels/chan_zap.c: Make sure we turn on the DSP when we answer
+ the call
+
+2007-10-26 22:21 +0000 [r87217] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES: Forgot to update CHANGES when I committed the linear
+ queue strategy. Thank you Russell, for pointing this out!
+
+2007-10-26 21:37 +0000 [r87202] Jason Parker <jparker@digium.com>
+
+ * channels/chan_local.c, channels/chan_zap.c,
+ channels/chan_agent.c, channels/chan_features.c,
+ res/res_crypto.c, res/res_realtime.c, res/res_monitor.c:
+ Correctly use defined return values in (some) load_module
+ functions. (issue #11096) Patches: chan_agent.c.patch uploaded by
+ eliel (license 64) chan_local.c.patch uploaded by eliel (license
+ 64) chan_features.c.patch uploaded by eliel (license 64)
+ chan_zap.c.patch uploaded by eliel (license 64)
+ res_monitor.c.patch uploaded by eliel (license 64)
+ res_realtime.c.patch uploaded by eliel (license 64)
+ res_crypto.c.patch uploaded by eliel (license 64)
+
+2007-10-26 17:39 +0000 [r87187] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c,
+ res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c,
+ res/ael/ael.tab.h, utils/ael_main.c,
+ pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex,
+ utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged
+ revisions 87168 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1
+ line closes issue #11086 where a user complains that references
+ to following contexts report a problem; The problem was REALLy
+ that he was referring to empty contexts, which were being
+ ignored. Reporter stated that empty contexts should be OK. I
+ checked it out against extensions.conf, and sure enough, empty
+ contexts ARE ok. So, I removed the restriction from AEL. This,
+ though, highlighted a problem with multiple contexts of the same
+ name. This should be OK, also. So, I added the extend keyword to
+ AEL, and it can preceed the 'context' keyword (mixed with
+ 'abstract', if nec.). This will turn off the warnings in AEL if
+ the same context name is used 2 or more times. Also, I now call
+ ast_context_find_or_create for contexts now, instead of just
+ ast_context_create; I did this because pbx_config does this. The
+ 'extend' keyword thus becomes a statement of intent. AEL can now
+ duplicate the behavior of pbx_config, ........
+
+2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Added queue
+ strategy "linear". This strategy is useful for those who always
+ wish for their phones to be rung in a specific order. (closes
+ issue #7279, reported and initially patched by diLLec, patch
+ reworked by me)
+
+ * configs/queues.conf.sample: Remove information about the
+ roundrobin strategy from trunk's queues.conf.sample since it no
+ longer exists
+
+2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007)
+ | 7 lines The addition of autoservice to func_curl additionally
+ made func_curl dependent on the existence of a channel, with no
+ real reason. This should make func_curl once again work without a
+ channel. Reported by jmls. Fixed by tilghman. Closes issue #11090
+ ........
+
+ * /: Blocked revisions 87067 via svnmerge ........ r87067 |
+ tilghman | 2007-10-25 17:53:06 -0500 (Thu, 25 Oct 2007) | 4 lines
+ Backport alternate encoding of newline delimiters from trunk to
+ 1.4, as approved by Russell Reported by blitzrage Closes issue
+ #10903 ........
+
+ * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c,
+ main/app.c: Use the same delimited character as the FILTER
+ function in FIELDQTY and CUT.
+
+2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /, include/asterisk/linkedlists.h: Merged
+ revisions 87069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007)
+ | 2 lines appending one list to another should leave the first
+ list empty, and not require the user to do that ........
+
+2007-10-25 18:59 +0000 [r87040] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Add support for a muted user to request to
+ talk. The '2' option in the user menu will adjust this status if
+ a user is muted. The talk request status will be reflected in the
+ CLI commands as well as the manager interface. (closes issue
+ #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded
+ by imesper (license 275)
+
+2007-10-25 16:21 +0000 [r87024] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c:
+ closes issue #11045 - each file needs to define
+ ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the
+ compiler flags; the problem was that the fixes were getting made
+ to the generated .c file, and erased the next time someone
+ regenerated that file from the corresponding .y or .flex file.
+ Moral of story: keep your eyes open and make mods to the .y (or
+ flex input file) and re-run bison (or flex) as the Makefile
+ directs for that file, and then check in both. Also,
+ res_config_sqlite was kinda missed, and has the same issue.
+
+2007-10-24 21:26 +0000 [r86985] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Adding the general
+ option "shared_lastcall" to queues so that a member's wrapuptime
+ may be used across multiple queues. (closes issue #9777, reported
+ and patched by eliel)
+
+2007-10-24 20:59 +0000 [r86983] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500
+ (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname
+ configuration option. Simplify code slightly in the process.
+ Issue 11079, reported by ddv2005 ........
+
+2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2,
+ pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c,
+ pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue
+ #11005, where #include uses the current dir instead of the config
+ dir (/etc/asterisk) for relative path includes for AEL
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1
+ line closes issue #11037 -- unable to specify app:spec in hint
+ arguments ........
+
+ * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1
+ line closes issue #11052 -- where nothing after the ? will allow
+ un-initialized variable values to corrupt and crash asterisk on
+ 64-bit platforms ........
+
+ * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1
+ line This should get rid of a really, really irritating warning
+ generated by some 64-bit platforms from libc, where free(0) is
+ frowned upon ........
+
+ * /, main/Makefile: Merged revisions 86881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1
+ line this update to Makefile corrects how ast_expr2f.c should be
+ generated ........
+
+2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22
+ Oct 2007) | 9 lines If lock tracking is not enabled, then we can
+ not attempt to log any mutex failures. If so, we could end up in
+ infinite recursion. The only lock that is affected by this is a
+ mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue
+ #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
+ (license 281) ........
+
+ * apps/app_playback.c: Convert some spaces to tabs and make it so
+ the CLI command is only registered once instead of 3 times.
+ (closes issue #11053) Reported by: seanbright Patches:
+ app_playback.patch uploaded by seanbright (license 71)
+
+2007-10-22 20:05 +0000 [r86820] Jason Parker <jparker@digium.com>
+
+ * main/udptl.c, channels/chan_local.c, main/frame.c,
+ res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
+ main/astobj2.c, main/config.c, main/cli.c,
+ channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
+ channels/chan_alsa.c, main/db.c, main/pbx.c,
+ channels/chan_agent.c, channels/iax2-provision.c,
+ apps/app_playback.c, channels/chan_misdn.c,
+ channels/chan_features.c, res/res_indications.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
+ res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
+ main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
+ res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
+ main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
+ res/res_agi.c, apps/app_minivm.c, main/logger.c,
+ res/res_realtime.c, main/image.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, res/res_clioriginate.c,
+ res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
+ channels/chan_sip.c, res/res_limit.c, main/translate.c,
+ res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
+ apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
+ channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
+ channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
+ funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to
+ AST_CLI_DEFINE, since the former didn't make much sense
+
+2007-10-22 17:40 +0000 [r86790] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astmm.c: Merged revisions 86787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007)
+ | 2 lines Minor FreeBSD build fix ........
+
+2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4
+ lines After reading online I have confirmed that Record-Route
+ headers should be copied to 1xx responses as well. (closes issue
+ #10113) Reported by: makoto ........
+
+ * /, apps/app_controlplayback.c: Merged revisions 86754 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4
+ lines Make sure res is a positive value before performing the
+ check to determine whether the user stopped it or not. (closes
+ issue #11023) Reported by: cfc ........
+
+2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 86750 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) |
+ 8 lines Don't leak a frame in the case that an END frame is
+ received and the time since the BEGIN is less than that of the
+ defined minimum DTMF duration. (closes issue #11051) Reported by:
+ casper Patches: channel.c.86664.diff uploaded by casper (license
+ 55) ........
+
+ * channels/chan_zap.c: There is a really fun game that you can play
+ before committing code, and it's called "make". :)
+
+ * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22
+ Oct 2007) | 4 lines Update the static mutex initializer to
+ include the initialization of the internal mutex used to protect
+ the lock debugging data. (closes issue #11044, patch suggested by
+ Ivan) ........
+
+2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval
+ defaulting to something other than 'never' doesn't seem to
+ accomplish any good and causes problems for plenty of people...
+
+2007-10-22 14:58 +0000 [r86696] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct
+ 2007) | 5 lines Account for the fact that sometimes headers may
+ be terminated with \r\n instead of just \n (closes issue #11043,
+ reported by yehavi) ........
+
+2007-10-22 14:56 +0000 [r86695] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/loader.c: merging patches that don't compile is bad...
+ mmkay?
+
+2007-10-22 14:28 +0000 [r86631-86664] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 86663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6
+ lines Move log message to before the frame it references is
+ freed. (closes issue #11050) Reported by: slavon Patches:
+ channel.c.86662.diff uploaded by casper (license 55) ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6
+ lines Fix tab completion for dundi show peer. (closes issue
+ #11041) Reported by: jsmith Patches:
+ asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
+ (license 176) ........
+
+ * /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct
+ 2007) | 6 lines Fixes for building under OpenSolaris. (closes
+ issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
+ uploaded by snuffy (license 35) ........
+
+2007-10-22 10:18 +0000 [r86616-86617] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
+ revisions 86598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) |
+ 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if
+ the dialplan does not match after an overlap call. Also added
+ out_cause=1 ........
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ started to add some basic support for supplementary services like
+ CallForwarding and so forth
+
+2007-10-21 22:52 +0000 [r86585] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged
+ revisions 85532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) |
+ 8 lines Properly handle the case where read() may return the text
+ for more than one CLI command at once for a remote console.
+ (closes issue #10888) Reported by: jamesgolovich Patches:
+ asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
+ 176) ........
+
+2007-10-20 19:56 +0000 [r86572] Matthew Fredrickson <creslin@digium.com>
+
+ * configs/zapata.conf.sample: Improved comments and organization
+ for zapata.conf (#10904)
+
+2007-10-19 18:46 +0000 [r86549] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add better support for blocking and
+ unblocking of CICs (#10965)
+
+2007-10-19 18:29 +0000 [r86534-86536] Jason Parker <jparker@digium.com>
+
+ * main/udptl.c, channels/chan_local.c, main/frame.c,
+ res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
+ main/astobj2.c, main/config.c, main/cli.c,
+ channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
+ channels/chan_alsa.c, main/db.c, main/pbx.c,
+ channels/chan_agent.c, channels/iax2-provision.c,
+ apps/app_playback.c, channels/chan_misdn.c,
+ channels/chan_features.c, res/res_indications.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
+ res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
+ main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
+ res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
+ main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
+ res/res_agi.c, apps/app_minivm.c, main/logger.c,
+ res/res_realtime.c, main/image.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, res/res_clioriginate.c,
+ res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
+ channels/chan_sip.c, res/res_limit.c, main/translate.c,
+ res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
+ apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
+ channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
+ channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
+ funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue
+ #11039, as suggested by seanbright.
+
+ * channels/chan_usbradio.c, res/res_config_pgsql.c,
+ channels/chan_misdn.c, channels/chan_h323.c,
+ res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c,
+ res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes
+ to NEW_CLI. Also fixes a few cli messages and some minor
+ formatting. (closes issue #11001) Reported by: seanbright
+ Patches: newcli.1.patch uploaded by seanbright (license 71)
+ newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch
+ uploaded by seanbright (license 71) newcli.5.patch uploaded by
+ seanbright (license 71) newcli.6.patch uploaded by seanbright
+ (license 71) newcli.7.patch uploaded by seanbright (license 71)
+
+2007-10-19 16:40 +0000 [r86470-86503] Joshua Colp <jcolp@digium.com>
+
+ * /, main/app.c: Merged revisions 86502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4
+ lines When returning a DTMF digit from ast_control_streamfile
+ cast it as a char so that 0 does not overlap with the success
+ return code. (closes issue #11023) Reported by: cfc ........
+
+ * /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6
+ lines Fix two issues with domains and transfers. If a port was
+ given in the hostname it was treated as part of the hostname. If
+ domains were configured but external domains were not enabled all
+ transfers would be considered remote. (closes issue #11027)
+ Reported by: ramonpeek Patches: 11027-1.diff uploaded by
+ ramonpeek (license 266) ........
+
+ * /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4
+ lines Set port number in received as information for
+ registrations as well. (closes issue #11028) Reported by: brad-x
+ ........
+
+2007-10-19 01:56 +0000 [r86439] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed a buffer size issue.
+
+2007-10-18 22:03 +0000 [r86407-86408] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 86405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500
+ (Thu, 18 Oct 2007) | 4 lines Add documentation for options in
+ asterisk.conf Issue 11029, patch by eserra ........
+
+ * /: Blocked revisions 86406 via svnmerge ........ r86406 | qwell |
+ 2007-10-18 17:01:02 -0500 (Thu, 18 Oct 2007) | 1 line Correct
+ documentation. I removed the wrong line.. ........
+
+2007-10-18 21:19 +0000 [r86373] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 86371-86372 via svnmerge ........ r86371 |
+ russell | 2007-10-18 16:14:15 -0500 (Thu, 18 Oct 2007) | 2 lines
+ Add support for setting the maximum trunk size for IAX2 trunking
+ ........ r86372 | russell | 2007-10-18 16:16:47 -0500 (Thu, 18
+ Oct 2007) | 2 lines Revert erroneous commit. ........
+
+2007-10-18 18:40 +0000 [r86350] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Fixing a segfault from tab-completing a "zap
+ restart" CLI command. (patch made by seanbright, pointed out in
+ #asterisk-dev on IRC)
+
+2007-10-18 18:06 +0000 [r86331] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /, include/asterisk/channel.h: Merged revisions
+ 86330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) |
+ 10 lines The channel needs to stay locked while running timer
+ callbacks, as they access and modify channel data that may change
+ elsewhere. I went through every timer callback in the source tree
+ to make sure that none of them did any additional locking that
+ could introduce deadlocks, and all is well. (closes issue #10765)
+ Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff
+ uploaded by Ivan (license 229) ........
+
+2007-10-18 17:40 +0000 [r86298-86329] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 86328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct
+ 2007) | 5 lines If a non-existent file is specified to be played
+ either as a periodic announcement or as a hold/position
+ announcement, the caller would be kicked out of the queue. No
+ longer does this happen. ........
+
+ * apps/app_queue.c: Changed some spaces to tabs
+
+2007-10-18 15:57 +0000 [r86297] Russell Bryant <russell@digium.com>
+
+ * /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) |
+ 3 lines Execute the RELEASE operation on transcoder channels in
+ the destroy callback. (patch from jsloan) ........
+
+2007-10-18 07:23 +0000 [r86277-86278] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue
+ #10784
+
+ * res/res_musiconhold.c: On reload, re-read the files in the
+ specified moh directory (closes issue #10536)
+
+2007-10-18 04:41 +0000 [r86238] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 86237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) |
+ 9 lines Revert a change that I made for issue #10979 which, as
+ has been pointed out to me in issue #11018, doesn't really make
+ sense. There is no reason to have the base64 decode function
+ force a '\0' terminated buffer, when the result is almost always
+ binary, anyway. In fact, this caused some breakage, as some code
+ in res_crypto passed in a buffer exactly the right size to get
+ its binary result, which got stomped on by this patch. (closes
+ issue #11018, reported by dimas) ........
+
+2007-10-17 21:41 +0000 [r86208] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 86202 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct
+ 2007) | 6 lines Changing the strategy field of the call_queue
+ struct to be signed instead of unsigned, since the code attempts
+ to set the strategy to -1 if you specify a bogus strategy. While
+ this isn't a huge issue in 1.4, it could be a problem for someone
+ who, say, tries to use the roundrobin strategy in trunk (despite
+ all the deprecation warnings in 1.4). ........
+
+2007-10-17 21:16 +0000 [r86195-86197] Tilghman Lesher <tlesher@digium.com>
+
+ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify
+ some preprocessor logic by using #elif
+
+ * CHANGES, configs/meetme.conf.sample: Document the changes made
+ earlier today to meetme
+
+2007-10-17 20:06 +0000 [r86180-86182] Steve Murphy <murf@digium.com>
+
+ * utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and
+ then, I noticed the clicompat stuff.
+
+ * utils/check_expr.c: more stub routines to allow linkage in
+ stand-alone environment, with thread debugs turned on
+
+ * utils/hashtest2.c: more stub routines to allow linkage in
+ stand-alone environment, with thread debugs turned on
+
+2007-10-17 18:01 +0000 [r86150] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) |
+ 8 lines If Asterisk is in the middle of shutting down, respond to
+ OPTIONS with 503 Unavailable. (closes issue #10994) Reported by:
+ eserra Patches: sip-options-503.patch uploaded by eserra (license
+ 45) ........
+
+2007-10-17 17:06 +0000 [r86119] Tilghman Lesher <tlesher@digium.com>
+
+ * main/term.c: Support color on certain platforms, even when
+ started at boot (before TERM is set) Closes issue #9048
+
+2007-10-17 17:00 +0000 [r86118] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4
+ lines Whoops, forgot to remove the original sip_scheddestroy.
+ (closes issue #11010) Reported by: vadim ........
+
+2007-10-17 16:09 +0000 [r86104] Jason Parker <jparker@digium.com>
+
+ * channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow
+ chan_usbradio to compile again. Closes issue #11014, patch by
+ seanbright.
+
+2007-10-17 15:39 +0000 [r86079] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 86066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007)
+ | 3 lines When runuser/rungroup is specified, a remote console
+ could only be attained by root (Closes issue #9999) ........
+
+2007-10-17 15:30 +0000 [r86067] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_usbradio.c: Change dependency for chan_usbradio to
+ asound. Let's keep everything uniform. (closes issue #11013)
+ Reported by: seanbright
+
+2007-10-17 15:13 +0000 [r86065] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Enhancements to realtime (closes issue #9609)
+
+2007-10-17 15:09 +0000 [r86064] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4
+ lines Don't schedule dialog destruction if a MESSAGE is received
+ using an existing dialog. (closes issue #11010) Reported by:
+ vadim ........
+
+2007-10-16 23:36 +0000 [r86029-86033] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 86032 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct
+ 2007) | 3 lines Since monitor-join is deprecated now, remove the
+ example from the sample queues.conf file ........
+
+ * apps/app_queue.c: Removed the monitor-join option. If one wishes
+ to mix audio, they should instead use monitor-type=mixmonitor.
+ (related to issue #10885)
+
+ * /: Blocked revisions 86028 via svnmerge ........ r86028 |
+ mmichelson | 2007-10-16 17:49:10 -0500 (Tue, 16 Oct 2007) | 6
+ lines Adding deprecated warning to monitor-join option, since the
+ plan is to no longer support this in favor of monitor-type =
+ mixmonitor (related to issue #10885) ........
+
+2007-10-16 22:36 +0000 [r85995-85998] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16
+ Oct 2007) | 1 line really picky formatting tweak ... ........
+
+ * /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16
+ Oct 2007) | 16 lines Some locking errors exposed the fact that
+ the lock debugging code itself was not thread safe. How ironic!
+ Anyway, these changes ensure that the code that is accessing the
+ lock debugging data is thread-safe. Many thanks to Ivan for
+ finding and fixing the core issue here, and also thanks to those
+ that tested the patch and provided test results. (closes issue
+ #10571) (closes issue #10886) (closes issue #10875) (might close
+ some others, as well ...) Patches: (from issue #10571)
+ ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
+ 229) - a few small changes by me ........
+
+2007-10-16 21:51 +0000 [r85959-85992] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fixing the build.
+
+ * apps/app_read.c: Fixing app_read so that if a timeout of less
+ than 1 ms is specified, assume that 1 ms is desired. (closes
+ issue #11000, reported and patched by michael-fig, with a warning
+ line added by me)
+
+ * /, apps/app_queue.c: Merged revisions 85958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct
+ 2007) | 5 lines Trying to remove a non-dynamic queue member via
+ dynamic means can lead to some interesting (read nasty)
+ situations. This patch clears up the issue by making only dynamic
+ queue members removable via dynamic methods. ........
+
+2007-10-16 20:55 +0000 [r85957] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Don't hangup the call for SS7 if we get an
+ alarm
+
+2007-10-16 20:32 +0000 [r85944] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't
+ improperly memset() over an ast_str. This was leftover from
+ before it got changed to use ast_str. (closes issue #11003,
+ reported by pj) (closes issue #10770, reported by yehavi)
+ (patched by me)
+
+2007-10-16 19:47 +0000 [r85943] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16
+ Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z
+ gnu extension to strftime) Reported and fixed by jcmoore Closes
+ issue #11002 ........
+
+2007-10-16 19:12 +0000 [r85897] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) |
+ 2 lines Remove a pointless lock. ........
+
+2007-10-16 16:40 +0000 [r85853-85883] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix IMAP compilation error. (closes issue
+ #10986, reported and patched by snuffy)
+
+ * /: Blocking changes from previous commit
+
+2007-10-16 15:15 +0000 [r85819-85851] Joshua Colp <jcolp@digium.com>
+
+ * /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4
+ lines Check to make sure a value has been given to the VMCOUNT
+ dialplan function. (closes issue #10996) Reported by: marsosa
+ ........
+
+ * main/threadstorage.c: Permit building under DEBUG_THREADLOCALS.
+ Thanks snuff.
+
+ * /, main/threadstorage.c: Merged revisions 85818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6
+ lines Fix memory allocation issue in threadstorage. (closes issue
+ #10995) Reported by: snuffy Patches: new-patch.diff uploaded by
+ snuffy (license 35) ........
+
+2007-10-16 10:38 +0000 [r85777-85787] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help
+ output
+
+ * channels/chan_jingle.c: Added two CLI functions, taken from
+ chan_gtalk : - jingle reload ; - jingle show channels.
+
+ * channels/chan_jingle.c: Make an audio path under the following
+ call configuration : SIP Phone 1 --- [chan_sip]Asterisk
+ 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP
+ Phone 2 Modifications : - set bridge type to partial ; - process
+ media candidates from the remote peer properly. Now we have
+ Jingle audio, at least between two Asterisk Jingle clients.
+
+2007-10-15 23:20 +0000 [r85764] Jason Parker <jparker@digium.com>
+
+ * configs/dundi.conf.sample, channels/chan_sip.c,
+ channels/chan_h323.c, main/acl.c, UPGRADE.txt,
+ channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c,
+ channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new
+ tos config format. Remove old unused defines for old style.
+ Closes issue 10860, patch by IgorG.
+
+2007-10-15 22:03 +0000 [r85751] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 85687 via svnmerge ........ r85687 |
+ tilghman | 2007-10-15 15:29:35 -0500 (Mon, 15 Oct 2007) | 5 lines
+ Don't execute a gosub if the arguments is zero-len (not just
+ NULL) Reported by davevg Fixed by me Closes issue #10985 ........
+
+2007-10-15 21:11 +0000 [r85718-85721] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 85720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) |
+ 3 lines Ensure that no pending state changes are leaked when the
+ device state change thread gets stopped on module unload.
+ ........
+
+ * /: Blocked revisions 85717 via svnmerge ........ r85717 | russell
+ | 2007-10-15 15:59:27 -0500 (Mon, 15 Oct 2007) | 7 lines
+ Previously, app_queue created a thread to handle every single
+ device state change. I changed this a while ago in trunk for
+ performance reasons. However, bug 8407 points out that it is
+ actually a race condition, causing device state changes to get
+ processed in random order. So, I backported my changes from trunk
+ to 1.4. (closes issue #8407, patch provided by tim_ringenbach,
+ committed patch by me) ........
+
+ * /, main/say.c: Merged revisions 85686 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) |
+ 7 lines Add a small fix for the tw version of saying dates.
+ (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
+ uploaded by sharkey (license 172) ........
+
+2007-10-15 20:16 +0000 [r85685] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 85684 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500
+ (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config'
+ target. Do not try to run chkconfig or similar if using DESTDIR.
+ Issue 10938, patch by cabal95. ........
+
+2007-10-15 20:09 +0000 [r85648-85683] Russell Bryant <russell@digium.com>
+
+ * doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the
+ channel var docs
+
+ * res/res_features.c, CHANGES: Added support for reading the
+ TOUCH_MONITOR_PREFIX channel variable. It allows you to configure
+ a prefix for auto-monitor recordings. (closes issue #6353)
+ Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded
+ by ivanfm (original patch) - updated patch:
+ 6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
+
+ * /, main/utils.c: Merged revisions 85649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) |
+ 2 lines Be pedantic about handling memory allocation failure.
+ ........
+
+ * /, main/utils.c: Merged revisions 85647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) |
+ 5 lines The loop in the handler for the "core show locks" could
+ potentially block for some amount of time. Be a little bit more
+ careful and prepare all of the output in an intermediary buffer
+ while holding a global resource. Then, after releasing it, send
+ the output to ast_cli(). ........
+
+2007-10-15 17:51 +0000 [r85633] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Document my changes from Friday
+
+2007-10-15 16:59 +0000 [r85605] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) |
+ 6 lines Make the default for the srvlookup option to be yes. It
+ doesn't really make sense for it to default to off. The default
+ configuration file has it on, and proper RFC behavior, as
+ indicated by a comment in the code, is for it to be on. So, let's
+ have it on by default to make lives easier. (closes issue #10954,
+ suggested by jtodd) ........
+
+2007-10-15 16:41 +0000 [r85578] Joshua Colp <jcolp@digium.com>
+
+ * /, configs/features.conf.sample: Merged revisions 85571 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4
+ lines Document that DTMF based features only work when two
+ channels are bridged together. (closes issue #10773) Reported by:
+ pbayley ........
+
+2007-10-15 16:36 +0000 [r85562] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/strings.h: Merged revisions 85561 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) |
+ 4 lines Make a few changes so that characters in the upper half
+ of the ISO-8859-1 character set don't get stripped when reading
+ configuration. (closes issue #10982, dandre) ........
+
+2007-10-15 16:23 +0000 [r85560] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 85559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4
+ lines Bring both DTMF begin and end frames up through to the core
+ for DTMF feature handling. (closes issue #10826) Reported by:
+ dimas ........
+
+2007-10-15 15:55 +0000 [r85557-85558] Russell Bryant <russell@digium.com>
+
+ * pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c.
+ This also makes the code a bit safer by removing various
+ assumptions about sizes. (No vulnerabilities, though) (closes
+ issue #10977) Reported by: dimas Patches: dundiparser.patch
+ uploaded by dimas (license 88)
+
+ * /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) |
+ 9 lines Ensure the buffer passed to ast_canmatch_extension() is
+ properly initialized so that it is null terminated. (issue
+ #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by
+ dimas (license 88) - small mods by me ........
+
+2007-10-15 15:26 +0000 [r85555] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c: Allow RTP structure registration
+
+2007-10-15 15:07 +0000 [r85553-85554] Joshua Colp <jcolp@digium.com>
+
+ * main/frame.c: Add packetization data for G.722. (closes issue
+ #10900) Reported by: andrew Patches: frame.diff uploaded by
+ andrew (license 240)
+
+ * /, main/rtp.c: Merged revisions 85552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4
+ lines If Monitor or a spy was added to a P2P or native bridged
+ channel bring the channel back to the generic bridging core so
+ the monitor or spy operations work. (closes issue #10943)
+ Reported by: julianjm ........
+
+2007-10-15 13:51 +0000 [r85551] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Allocate more space for the base64 output we
+ need to generate. Closes issue #10913, reported by tootai, who
+ graciously granted us access to his Asterisk server, thanks!
+ Daniel, feel free to reopen the bug in case you can reproduce
+ this on 1.4.
+
+2007-10-15 13:44 +0000 [r85539-85550] Russell Bryant <russell@digium.com>
+
+ * main/cli.c: Move the CLI commands that were in builtins[] into
+ the cli_cli[] array of CLI commands and remove the cli_iterator
+ struct. This gets tab completion working again. (closes issue
+ #10970) Reported by: jamesgolovich Patches:
+ asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license
+ 176)
+
+ * /: Blocked revisions 85548 via svnmerge ........ r85548 | russell
+ | 2007-10-15 08:16:23 -0500 (Mon, 15 Oct 2007) | 7 lines Suppress
+ a LOG_DEBUG message if debug is not enabled. (closes issue
+ #10980) Reported by: casper Patches: db.c.84633.diff uploaded by
+ casper (license 55) ........
+
+ * doc/tex/jitterbuffer.tex, doc/tex/extensions.tex,
+ doc/tex/channelvariables.tex, doc/tex/ael.tex,
+ doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex,
+ doc/tex/dundi.tex, doc/tex/security.tex,
+ doc/tex/configuration.tex, doc/tex/ajam.tex,
+ doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex,
+ doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex,
+ doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c,
+ doc/tex/localchannel.tex, doc/tex/cdrdriver.tex,
+ doc/tex/queuelog.tex: Another major doc directory update from
+ IgorG. This patch includes - Many uses of the astlisting
+ environment around verbatim text to ensure that it gets properly
+ formatted and doesn't run off the page. - Update some things that
+ have been deprecated. - Add escaping as needed - and more ...
+ (closes issue #10978) Reported by: IgorG Patches:
+ texdoc-85542-1.patch uploaded by IgorG (license 20)
+
+ * /, main/asterisk.c: Merged revisions 85545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) |
+ 7 lines Make sure remote consoles unmute themselves again after
+ reconnecting. (closes issue #10847) Reported by: atis Patches:
+ console_unmute_on_reconnect.patch uploaded by atis (license 242)
+ ........
+
+ * /, main/utils.c: Merged revisions 85543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) |
+ 8 lines Make sure that the base64 decoder returns a terminated
+ string. (closes issue #10979) Reported by: ys Patches:
+ util.c.diff uploaded by ys (license 281) - small mods by me
+ ........
+
+ * configure, configure.ac: Change the configure script to check for
+ a function that was recently added to libss7.
+
+ * /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) |
+ 7 lines Don't create the context for users in users.conf until we
+ know at least one user exists. (closes issue #10971) Reported by:
+ dimas Patches: pbxconfig.patch uploaded by dimas (license 88)
+ ........
+
+ * doc/tex/backtrace.tex (added): When merging the last
+ documentation update, I forgot to "svn add" a file. Here it is.
+ (closes issue #10962)
+
+2007-10-13 15:34 +0000 [r85537] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 85536 via svnmerge ........ r85536 |
+ tilghman | 2007-10-13 10:26:01 -0500 (Sat, 13 Oct 2007) | 4 lines
+ Remove deprecated syntax from sample ael file Reported and
+ patched by: dimas Closes issue #10967 ........
+
+2007-10-13 08:38 +0000 [r85535] James Golovich <james@gnuinter.net>
+
+ * main/cli.c: Fix compiling cli.c due to differences with new cli
+ system (closes issue 0010966)
+
+2007-10-13 05:53 +0000 [r85534] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged
+ revisions 85533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) |
+ 12 lines Fix an issue with console verbosity when running
+ asterisk -rx to execute a command and retrieve its output. The
+ issue was that there was no way for the main Asterisk process to
+ know that the remote console was connecting in the -rx mode. The
+ way that James has fixed this is to have all remote consoles
+ muted by default. Then, regular remote consoles automatically
+ execute a CLI command to unmute themselves when they first start
+ up. (closes issue #10847) Reported by: atis Patches:
+ asterisk-consolemute.diff.txt uploaded by jamesgolovich (license
+ 176) ........
+
+2007-10-12 20:06 +0000 [r85527] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Allow for the
+ position announcement to be turned off if desired. (closes issue
+ #8515, reported by bruno_rocha, initial patch by bruno_rocha,
+ final patch by qwell)
+
+2007-10-12 19:41 +0000 [r85525-85526] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to
+ finish the last of the charge_number patch up #10916
+
+ * channels/chan_zap.c: Add support for receive charge number in
+ dialplan #10916
+
+2007-10-12 18:37 +0000 [r85522-85524] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions
+ 85523 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007)
+ | 2 lines Change Digium address ........
+
+ * funcs/func_strings.c: Enable ranges, hexadecimal, octal, and
+ special backslashed characters for the FILTER function
+
+2007-10-12 15:50 +0000 [r85516-85519] Russell Bryant <russell@digium.com>
+
+ * doc/tex/odbcstorage.tex, doc/tex/extensions.tex,
+ doc/tex/channelvariables.tex, doc/tex/ael.tex,
+ doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex,
+ doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex,
+ doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex,
+ doc/tex/localchannel.tex, doc/tex/ices.tex,
+ doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory
+ improvements, including: - Added development section
+ (backtrace.tex) - Correct filesystem path formating - Replace all
+ "|" argument separator to "," - Endless count of spaces at the
+ end of line - Using astlisting to make listings do not take so
+ much place - Take back ASTRISKVERSION on first page - Make
+ localchannel.tex readable by inserting extra end of lines (closes
+ issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch
+ uploaded by IgorG (license 20)
+
+ * res/res_smdi.c, /: Merged revisions 85517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) |
+ 3 lines Fix a spelling error in a log message. SMDI, not SDMI.
+ (closes issue #10959) ........
+
+ * /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) |
+ 7 lines Fix the potential use of an uninitialized buffer in a log
+ message. (closes issue #10958) Reported by: dimas Patches:
+ realtime.patch uploaded by dimas (license 88) ........
+
+2007-10-11 22:42 +0000 [r85474-85499] Matthew Fredrickson <creslin@digium.com>
+
+ * apps/app_dial.c: Make sure we propogate ANI2 to the outbound
+ channel
+
+ * funcs/func_callerid.c: See if I can fix this borked ANI2 code I
+ added
+
+ * channels/chan_zap.c: Make sure we set the ANI2 field for PRI
+
+ * funcs/func_callerid.c: Add ANI2 support to func_callerid
+
+ * channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916
+
+ * channels/chan_zap.c: Add CCR test support #10916
+
+2007-10-11 19:03 +0000 [r85460] Russell Bryant <russell@digium.com>
+
+ * main/udptl.c, main/threadstorage.c, res/res_limit.c,
+ main/translate.c, res/res_crypto.c, res/res_convert.c,
+ channels/iax2-provision.c, channels/chan_gtalk.c,
+ channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c,
+ main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c,
+ pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c,
+ main/asterisk.c, main/db.c, channels/chan_mgcp.c,
+ res/res_clioriginate.c: Merge a ton of NEW_CLI conversions.
+ Thanks to everyone that helped out! :) (closes issue #10724)
+ Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel
+ (license 64) chan_oss.c.patch uploaded by eliel (license 64)
+ chan_mgcp.c.patch2 uploaded by eliel (license 64)
+ pbx_config.c.patch uploaded by seanbright (license 71)
+ iax2-provision.c.patch uploaded by eliel (license 64)
+ chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch
+ uploaded by seanbright (license 71) file.c.patch uploaded by
+ seanbright (license 71) image.c.patch uploaded by seanbright
+ (license 71) cli.c.patch uploaded by moy (license 222)
+ astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch
+ uploaded by moy (license 222) res_limit.c.patch uploaded by
+ seanbright (license 71) res_convert.c.patch uploaded by
+ seanbright (license 71) res_crypto.c.patch uploaded by seanbright
+ (license 71) app_osplookup.c.patch uploaded by seanbright
+ (license 71) app_rpt.c.patch uploaded by seanbright (license 71)
+ app_mixmonitor.c.patch uploaded by seanbright (license 71)
+ channel.c.patch uploaded by seanbright (license 71)
+ translate.c.patch uploaded by seanbright (license 71)
+ udptl.c.patch uploaded by seanbright (license 71)
+ threadstorage.c.patch uploaded by seanbright (license 71)
+ db.c.patch uploaded by seanbright (license 71) cdr.c.patch
+ uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy
+ (license 222) app_osplookup-rev83558.patch uploaded by moy
+ (license 222) res_clioriginate.c.patch uploaded by moy (license
+ 222)
+
+2007-10-11 17:17 +0000 [r85431-85444] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Let's hard code this until I fix it
+
+ * channels/chan_zap.c: Make sure we are clean to build without
+ libpri
+
+2007-10-11 15:29 +0000 [r85398] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 85397 via svnmerge ........ r85397 | file |
+ 2007-10-11 12:26:20 -0300 (Thu, 11 Oct 2007) | 6 lines When
+ creating a new packet don't try to stop retransmission of it. It
+ was just allocated/created so it's impossible for it to have
+ already been scheduled. (closes issue #10945) Reported by:
+ flefoll Patches: chan_sip.c.br14.85280.xmit_reliable-patch
+ uploaded by flefoll (license 244) ........
+
+2007-10-11 04:40 +0000 [r85357] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 85356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007)
+ | 2 lines A dollar sign by itself, not indicating a start of a
+ variable or expression prematurely ends substitution (closes
+ issue #10939) ........
+
+2007-10-10 16:01 +0000 [r85317] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/file.h, /: Merged revisions 85316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10
+ Oct 2007) | 6 lines I introduced a new member to the
+ ast_filestream struct in 1.4.12, but put it in the middle of the
+ struct, instead of at the end. One of the Debian folks, paravoid,
+ pointed out that this breaks binary compatability with modules
+ compiled against older headers. So, I'm moving the new member to
+ the end of the struct to resolve the situation. ........
+
+2007-10-10 14:43 +0000 [r85281] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4
+ lines If devicestate is passed a port number strip it out.
+ (closes issue #10930) Reported by: ibc ........
+
+2007-10-10 14:38 +0000 [r85279] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct
+ 2007) | 5 lines A bunch of changes from sprintf to snprintf. See
+ security advisory AST-2002-022 ........
+
+2007-10-10 14:30 +0000 [r85234-85278] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6
+ lines Add support for handling a 182 Queued response. (closes
+ issue #10924) Reported by: ramonpeek Patches: queued-182.diff
+ uploaded by ramonpeek (license 266) ........
+
+ * /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6
+ lines Close voicemail message description file if duration did
+ not meet the minimum, or else we will eventually run out of file
+ descriptors. (closes issue #10918) Reported by: brak2718 Patches:
+ vm1.4.12.1.patch uploaded by brak2718 (license 279) ........
+
+ * main/logger.c: Process outstanding log messages before shutting
+ down the logger thread. (closes issue #10933) Reported by:
+ sperreault
+
+2007-10-10 06:48 +0000 [r85197] Luigi Rizzo <rizzo@icir.org>
+
+ * bootstrap.sh: Adapt the autotools names to different versions of
+ FreeBSD (and open the way to better adaptation for other
+ platforms as well).
+
+2007-10-10 06:41 +0000 [r85196] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10
+ Oct 2007) | 2 lines use a macro instead of an inline function, so
+ that backtraces will report the caller of ast_frame_free()
+ properly ........
+
+2007-10-09 22:35 +0000 [r85177] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Patch to add one-touch parking for queues.
+ (closes issue #10869, reported and patched by bluecrow76)
+
+2007-10-09 22:21 +0000 [r85140-85176] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged
+ revisions 85158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007)
+ | 5 lines This commit fixes the following issues: - Deadlock in
+ ast_write (issue #10406) - Deadlock in ast_read (issue #10406) -
+ Possible mutex initialization error in lock.h (issue #10571)
+ ........
+
+ * apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c,
+ apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c,
+ utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove
+ redundant includes (patch by snuffy) (Closes issue #10922)
+
+2007-10-09 15:12 +0000 [r85097-85098] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Note jitterbuffer support for chan_local in CHANGES
+
+ * channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer
+ support for chan_local. To enable it, you use the 'j' option in
+ the Dial command. The 'j' option _must_ be used in conjunction
+ with the 'n' option. This feature will allow you to use the
+ existing jitterbuffer implementation to put a jitterbuffer on
+ incoming SIP calls connecting to Asterisk applications by putting
+ a local channel in the middle.
+
+2007-10-09 14:31 +0000 [r84991-85094] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4
+ lines Don't perform a reinvite if a transfer is in progress.
+ (issue #10915) Reported by: ramonpeek ........
+
+ * /, main/rtp.c: Merged revisions 85057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4
+ lines Only update codec information if the channel has a
+ technology private structure. (issue #10915) Reported by:
+ ramonpeek ........
+
+ * res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c,
+ main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it
+ compiles when MTX Profiling is enabled. (closes issue #10898)
+ Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by
+ qwell (license 4)
+
+ * /, main/rtp.c: Merged revisions 85023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4
+ lines Update codec information as well as address when doing hold
+ reinvites. (issue #10868) Reported by: mavince ........
+
+ * main/channel.c, /: Merged revisions 84990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4
+ lines Don't keep trying to native bridge if either of the
+ channels are involved in a masquerade operation to be done.
+ (closes issue #10696) Reported by: tbelder ........
+
+2007-10-08 03:29 +0000 [r84958] Russell Bryant <russell@digium.com>
+
+ * /, Makefile.rules: Merged revisions 84957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) |
+ 6 lines Enable file dependency tracking for _all_ builds, and not
+ just for builds with dev-mode enabled. I have seen enough
+ problems caused by this that I don't think it's worth keeping. I
+ want to continue to encourage anybody that is interested to
+ continue to run Asterisk from svn. Furthermore, I do not want
+ their systems to break when we change a structure definition in a
+ header file. :) ........
+
+2007-10-07 16:28 +0000 [r84891-84939] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * configs/jabber.conf.sample, include/asterisk/jabber.h,
+ res/res_jabber.c: Make the status and priority configurable.
+ Closes issue #10785, patch by Luke-Jr, thanks!
+
+ * /, res/res_jabber.c: Merged revisions 84902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007)
+ | 5 lines Presence packets from a client who's connected with our
+ Jabber ID are valid, therefore, those clients must be considered
+ as buddies. The resource string helps us make the distinction
+ between clients. Closes issue #10707, reported by yusufmotiwala.
+ ........
+
+ * res/res_jabber.c: Fix indentation
+
+ * /, res/res_jabber.c: Merged revisions 84890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007)
+ | 5 lines Prevent Asterisk from crashing when receiving a
+ presence packet without resource from a buddy that is known to
+ have a resource list. Revert a change I previously made, where
+ Asterisk could point to a freed memory location. ........
+
+2007-10-05 19:48 +0000 [r84852] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/db.c: Merged revisions 84851 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007)
+ | 2 lines Log exactly why we can't open the database, if we fail
+ (closes issue #10887) ........
+
+2007-10-05 18:57 +0000 [r84819] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 84818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4
+ lines Update the remembered RTP peer information when putting an
+ endpoint on hold or taking it off hold so that the RTP stack does
+ not initiate a needless reinvite. (closes issue #10868) Reported
+ by: mavince ........
+
+2007-10-05 16:49 +0000 [r84743-84784] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) |
+ 4 lines Do deadlock avoidance in a couple more places. You can't
+ lock two channels at the same time without doing extra work to
+ make sure it succeeds. (closes issue #10895, patch by me)
+ ........
+
+ * main/manager.c, /: Merged revisions 84742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) |
+ 3 lines Fix a copy/paste error in the description of UpdateConfig
+ that was pointed out by JerJer on #asterisk-dev ........
+
+2007-10-04 22:58 +0000 [r84693-84726] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix
+ some bad logic in the counting of statistics for QueueSummary
+ manager event. Variables were not being reset for each additional
+ queue, so cumulative totals were reported on each successive
+ queue. 2) Add a longest hold time stat to QueueSummary manager
+ event.
+
+ * /, apps/app_queue.c: Merged revisions 84692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct
+ 2007) | 5 lines Don't allocate space for queue members unless
+ it's needed. You end up deleting dynamic members on a reload. Not
+ good. closes issue (#10879, reported by dazza76, patched by me)
+ ........
+
+2007-10-04 21:38 +0000 [r84691] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007)
+ | 2 lines callers of sig2str already add the word 'signalling' in
+ the appropriate place, so don't duplicate it ........
+
+2007-10-04 16:56 +0000 [r84671] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c: Update to current coding standards, also
+ changing the argument delimiter to ',' (Closes issue #10876)
+
+2007-10-04 14:54 +0000 [r84613-84638] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 84637 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4
+ lines Create a duplicate of the channel's member name as the tab
+ completion stuff will free it. (closes issue #10884) Reported by:
+ adamg ........
+
+ * main/pbx.c: Don't register the exception function with module
+ information. Since it is in the core there is none and it will
+ explode.
+
+2007-10-03 23:05 +0000 [r84580-84582] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/rtp.c: Merged revisions 84581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007)
+ | 2 lines When an RFC 2833 event is sent that we don't recognize,
+ ignore it, don't queue a NULL digit (closes issue #10877)
+ ........
+
+ * main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h:
+ Create a universal exception handling extension, "e" (closes
+ issue #9785)
+
+2007-10-03 18:23 +0000 [r84512-84545] Steve Murphy <murf@digium.com>
+
+ * /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the
+ CUT in AEL
+
+ * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1
+ line closes issue #10834 ; where a null input to a switch
+ statement results in a hangup; since switch is implemented with
+ extensions, and the default case is implemented with a '.', and
+ the '.' matches 1 or more remaining characters, the case where 0
+ characters exist isn't matched, and the extension isn't matched,
+ and the goto fails, and a hangup occurs. Now, when a default case
+ is generated, it also generates a single fixed extension that
+ will match a null input. That extension just does a goto to the
+ default extension for that switch. I played with an alternate
+ solution, where I just tack an extra char onto all the patterns
+ and the goto, but not the default case's pattern. Then even a
+ null input will still have at least one char in it. But it made
+ me nervous, having that extra char in , even if that's a pretty
+ secret and low-level issue. ........
+
+2007-10-02 20:07 +0000 [r84475] Russell Bryant <russell@digium.com>
+
+ * Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) |
+ 5 lines * Don't build the menuselect-tree for the tarball, as it
+ requires running the configure script first * Change the Makefile
+ to note that menuselect-tree depends on the configure script.
+ ........
+
+2007-10-02 19:02 +0000 [r84432-84440] Jason Parker <jparker@digium.com>
+
+ * /: Blocked revisions 84437 via svnmerge ........ r84437 | qwell |
+ 2007-10-02 14:01:59 -0500 (Tue, 02 Oct 2007) | 1 line Fix some
+ odd formatting I missed.. ........
+
+ * /, res/res_features.c: Merged revisions 84410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500
+ (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel
+ before return on failure. Issue 10821, patch by Ivan ........
+
+2007-10-02 18:12 +0000 [r84405] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Add MSet for people who prefer the old, deprecated
+ syntax of Set (Closes issue #10549)
+
+2007-10-02 14:13 +0000 [r84371] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) |
+ 6 lines Use snprintf instead of sprintf in one place. There is no
+ vulnerability here due to various buffer sizes around the code,
+ but I still didn't like seeing a non length-limited copy of data
+ coming off of the wire into a stack buffer, as this would be a
+ problem in the future if buffer sizes elsewhere got changed or
+ size limitations removed ... ........
+
+2007-10-02 13:58 +0000 [r84368] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Don't swap channel priority if using epoll as polling
+ should/will only happen off the first channel. (closes issue
+ #10867) Reported by: phsultan
+
+2007-10-01 23:33 +0000 [r84327-84331] Steve Murphy <murf@digium.com>
+
+ * utils/check_expr.c: OK. THis a DEBUG_THREADS situation.
+
+ * utils/check_expr.c: picky gcc versions... sigh.
+
+ * utils/check_expr.c: This mod will allow check_expr to compile in
+ the presence of DEBUG_THREAD situations. At least, it does for
+ me. And it's less expensive than several other approaches I
+ tried.
+
+ * res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y,
+ pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1
+ line closes issue #10777 -- by returning a null for the parse
+ tree when there's really nothing there, and making sure we don't
+ try to do checking on a null tree. ........
+
+2007-10-01 21:54 +0000 [r84300] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, Makefile.rules, channels/Makefile: Merged revisions
+ 84291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6
+ lines Add dist-clean support for subdirs. Change h323 to only
+ remove the Makefile on a dist-clean, rather than a clean. This
+ fixes a bug I found with trying to run make after a make clean
+ ........
+
+2007-10-01 21:31 +0000 [r84275] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged
+ revisions 84274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007)
+ | 1 line moved get_base_channel() code from action_redirect to
+ ast_channel_masquerade() for issue 7706 and BE-160 ........
+
+2007-10-01 21:15 +0000 [r84207-84272] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) |
+ 4 lines Fulfull a feature request from Qwell on the "core show
+ locks" output. It will now note the lock type for each lock that
+ a thread holds. (mutex, rdlock, or wrlock) ........
+
+ * /, res/res_agi.c: Merged revisions 84236 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) |
+ 5 lines Add another sanity check in the AGI read loop. We really
+ don't care about EAGAIN unless we didn't read an entire line. If
+ there is a newline at the end if the read buffer, break, because
+ we got the whole thing. (reported and patched by bmd) ........
+
+ * /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01
+ Oct 2007) | 2 lines Show rwlocks in the "core show locks" output.
+ Before, it only showed mutexes. ........
+
+2007-10-01 15:57 +0000 [r84176] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Check to make sure a structure pointer is
+ non-NULL before touching it... crashing is bad, mmmk? (closes
+ issue #10831) Reported by: eliel Patches: chan_sip.c.patch
+ uploaded by eliel (license 64)
+
+2007-10-01 15:34 +0000 [r84167-84174] Russell Bryant <russell@digium.com>
+
+ * main/say.c: Change simple uses of snprintf to ast_copy_string.
+ This was provided by mvanbaak as a part of issue #10843, but this
+ part didn't apply because of a patch I applied right beforehand.
+
+ * channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c,
+ apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
+ main/say.c, apps/app_minivm.c, pbx/dundi-parser.c,
+ channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c,
+ main/rtp.c, channels/chan_mgcp.c: Corydon posted this janitor
+ project to the bug tracker and mvanbaak provided a patch for it.
+ It replaces a bunch of simple calls to snprintf with
+ ast_copy_string (closes issue #10843) Reported by: Corydon76
+ Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7)
+
+ * main/say.c: Simplify code by using the -= and %= operators.
+ (closes issue #10848) Reported by: opticron Patches: saymod.diff
+ uploaded by opticron (license 267)
+
+ * codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The
+ trunk version of this patch also includes a couple more small
+ clean fixes from IgorG. Merged revisions 84170 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) |
+ 3 lines Remove another file in "make clean". (closes issue
+ #10814, paravoid) ........
+
+ * main/cli.c: Don't set the full command string until after
+ verifying that there is not another CLI command with the same
+ command text registered. This prevents a crash if someone
+ accidentally calls ast_cli_register() on the same CLI command
+ data twice. This also fixes a small bug where the helpers list
+ would get unlocked without being locked if building the full
+ command failed. (closes issue #10858, reported by jamesgolovich,
+ patched by me)
+
+ * configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new
+ option for files-based music on hold to ensure that the sort
+ order of the files is alphabetical. (closes issue #10855)
+ Reported by: jamesgolovich Patches:
+ asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license
+ 176)
+
+ * apps/app_dial.c, /: Merged revisions 84166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) |
+ 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........
+
+2007-10-01 14:21 +0000 [r84159-84165] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Add MP4 to part of the SDP code. (closes
+ issue #10820) Reported by: ruikubo Patches: chan_sip.patch
+ uploaded by ruikubo (license 250)
+
+ * /: Blocked revisions 84163 via svnmerge ........ r84163 | file |
+ 2007-10-01 11:10:47 -0300 (Mon, 01 Oct 2007) | 4 lines Remove
+ chan_usbradio config file from tree, it is not present in here.
+ (closes issue #10839) Reported by: casper ........
+
+ * main/dnsmgr.c: Don't register the dnsmgr refresh CLI command
+ twice. (closes issue #10856) Reported by: jamesgolovich Patches:
+ asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license
+ 176)
+
+ * /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct
+ 2007) | 6 lines Fix randomness. save_pos was being set to 0
+ initially instead of -1, causing it to jump to position 0 when
+ moh started. (closes issue #10859) Reported by: jamesgolovich
+ Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
+ (license 176) ........
+
+ * apps/app_dial.c, /: Merged revisions 84158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4
+ lines Only attempt early bridging if the options given to Dial()
+ permit it. (closes issue #10861) Reported by: peekyb ........
+
+2007-09-30 20:06 +0000 [r84143-84147] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/module.h: Merged revisions 84146 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30
+ Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++
+ modules. The load and reload parameters were in the wrong place.
+ (closes issue #10846, alebm) ........
+
+ * funcs/func_lock.c: * The documentation for the LOCK() function
+ says that it will block for up to 3 seconds while waiting on a
+ lock when other locks are currently held to avoid deadlocks.
+ Change the code to reflect this. * Since trying to grab a lock
+ may block for some time, put the channel in autoservice so that
+ audio is still read from the channel and that any active
+ generators on the channel don't pause.
+
+2007-09-29 23:47 +0000 [r84134-84137] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1
+ line This issue sort of closes 10786; All config files support
+ #include with globbing (you know, *,[chars],?,{list,list},etc),
+ so I've updated the AEL system to support this also. ........
+
+ * pbx/ael/ael-test/ael-ntest22/t2 (added),
+ pbx/ael/ael-test/ael-ntest22/t3 (added),
+ pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
+ pbx/ael/ael-test/ael-ntest22 (added),
+ pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
+ (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
+ pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
+ pbx/ael/ael-test/ref.ael-test4,
+ pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
+ pbx/ael/ael-test/ael-ntest22/qq.ael (added),
+ pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL
+ affected a small number of tests. Added a regression test for
+ glob'd includes
+
+2007-09-29 18:21 +0000 [r84130] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_manager.c: Set enablecdr at the end of re-reading the
+ config file (Closes issue #10852)
+
+2007-09-29 00:19 +0000 [r84115] Matthew Fredrickson <creslin@digium.com>
+
+ * main/translate.c: Let's use process time instead of wall clock
+ time for show translation
+
+2007-09-28 14:35 +0000 [r84050-84080] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Autoconf requires version 2.60, not
+ 2.59, to process (Closes issue #10842)
+
+ * /, main/say.c: Merged revisions 84078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007)
+ | 2 lines Correct pronunciations of numbers for .nl (Closes issue
+ #10837) ........
+
+ * main/channel.c, /: Merged revisions 84049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007)
+ | 3 lines Avoid a deadlock with ALL of the locks in the
+ masquerade function, not just the pairs of channels. (Closes
+ issue #10406) ........
+
+2007-09-27 23:18 +0000 [r84019] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * main/manager.c, /, channels/chan_agent.c,
+ include/asterisk/channel.h: Merged revisions 84018 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27
+ Sep 2007) | 1 line if an Agent is redirected, the base channel
+ should actually be redirected. This was causing multiple issues,
+ especially issue 7706 and BE-160 ........
+
+2007-09-27 00:08 +0000 [r83978-83986] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007)
+ | 2 lines avoid the weird usage of assert() in the ALSA header
+ files that gcc 4.2 wants to complain about ........
+
+ * res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const
+ pointer warnings
+
+2007-09-27 00:02 +0000 [r83911-83977] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) |
+ 1 line remove a todo item that has been completed ........
+
+ * /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) |
+ 2 lines I changed my mind ... I think this should be a
+ LOG_NOTICE. ........
+
+ * /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) |
+ 5 lines Add a log message that was requested by the masses in the
+ developer tutorial session at Astricon. chan_sip did not output
+ any message when a call was rejected because the extension was
+ not found. This adds a verbose message (at verbose level 3) to
+ note when this happens. ........
+
+ * /: Merged revisions 83910 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) |
+ 3 lines Fix building chan_misdn under dev-mode. (please run the
+ configure script with --enable-dev-mode so this doesn't happen
+ again ...) ........
+
+2007-09-26 18:43 +0000 [r83880] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007)
+ | 2 lines Remove unused 4k of memory on the program stack (closes
+ issue #10827) ........
+
+2007-09-26 06:53 +0000 [r83849-83864] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h: fix a typo in a comment
+
+ * include/asterisk/file.h: Change function documentation to use
+ doxygen tags. (Really, I just needed to make some minor change in
+ trunk to test something with automerge ...)
+
+2007-09-25 23:14 +0000 [r83834] Matthew Fredrickson <creslin@digium.com>
+
+ * doc/ss7.txt: Fix typo in readme
+
+2007-09-25 21:06 +0000 [r83819] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h: Don't note that functions are
+ deprecated in favor of themselves. This was found by showing a
+ very poor example doxygen function in a presentation this
+ morning. :)
+
+2007-09-25 16:34 +0000 [r83804] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Added a CLI command that shows our buddy list,
+ as suggested by Daniel McKeehan, thanks!
+
+2007-09-25 14:18 +0000 [r83774] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 83773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007)
+ | 2 lines jmls pointed out that unsetting the group and setting
+ the group to the blank string aren't quite the same. ........
+
+2007-09-25 13:41 +0000 [r83758] Joshua Colp <jcolp@digium.com>
+
+ * res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a
+ value without freeing the previous result is bad, mmmk?
+
+2007-09-25 09:07 +0000 [r83743] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c, include/asterisk/jingle.h: Comply with
+ latest XEP-0166, XEP-0167, XEP-0176. No real Jingle
+ implementation being available, testing was made using two
+ Asterisk servers relaying SIP calls over their Jingle channels:
+ SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] ---
+ [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was
+ possible to test the code in both ways, and make the Jingle
+ channel comply with the latest specifications. No sound available
+ yet. Main modifications include : - modified the
+ 'jingle_candidate' structure and the 'jingle_create_candidates'
+ function according to XEP-0176 ; - modified the 'jingle_action'
+ function in order to properly terminate a Jingle session, in
+ conformance with XEP-0166 ; - modified username format used in
+ STUN requests ; - actually make the bindaddr configuration field
+ useable. Todo : - set audio paths up (no native bridging) ; -
+ make the CLI gtalk functions available to jingle ; - clean up the
+ storage space used in strings.
+
+2007-09-25 08:09 +0000 [r83741] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, utils: Add some files to the utils directory
+ svn:ignore and Makefile clean target (closes issue #10808,
+ reported by mvanbaak)
+
+2007-09-24 22:06 +0000 [r83696-83726] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, main/asterisk.c: Permit custom locations for astdb and
+ the keys directory (though default to the current locations)
+ (Closes issue #10267)
+
+ * /, build_tools/make_defaults_h: Merged revisions 83695 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007)
+ | 4 lines In the source, keys are relative to the datadir, not
+ varlib (which is the same in most cases, but it's good to be
+ accurate). Closes issue #10811 ........
+
+2007-09-24 17:10 +0000 [r83671] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's
+ patch for configurable SDP origin-field username and session
+ field, closes issue# 10795
+
+2007-09-24 17:00 +0000 [r83656] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: interface_exists_global was never returning 1.
+ Most likely an error from my merge on Friday. (closes issue
+ #10817, reported and patched by snar, patch simplified by me)
+
+2007-09-24 16:42 +0000 [r83654-83655] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 83653 via svnmerge ........ r83653 |
+ tilghman | 2007-09-24 11:37:52 -0500 (Mon, 24 Sep 2007) | 2 lines
+ Oops. Removed the unworkable workaround. This note should never
+ have been in the release. ........
+
+ * /, main/app.c: Merged revisions 83637 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007)
+ | 3 lines Making change to group splitting, as discussed on the
+ -dev list. The main effect of this will be to permit
+ Set(GROUP([cat])=), i.e. unsetting a group. ........
+
+2007-09-22 19:54 +0000 [r83575-83590] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 83589 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1
+ line This closes issue #10788 -- The exact same fixes are made
+ here for the first arg in the for(arg1; arg2; arg3) {} statement,
+ as were done for the 3rd arg. It can now be an assignment that
+ will embedded in a Set() app, or a macro call, or an app call.
+ ........
+
+ * res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1
+ line This closes issue #10788 -- the 3rd arg in the for statement
+ is now wrapped in Set() only if there's an '=' in that string.
+ Otherwise, if it begins with '&', then a Macro call is generated;
+ otherwise it is made into an app call. A bit more accomodating,
+ keeps the new guys happy, and the guys with ael-1 code should be
+ happy, too ........
+
+2007-09-22 17:37 +0000 [r83574] Matthew Fredrickson <creslin@digium.com>
+
+ * doc/ss7.txt: Fix potential point of confusion
+
+2007-09-22 14:45 +0000 [r83517-83545] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix
+ build of check_expr and hashtest2 when DEBUG_THREADLOCAL is
+ defined
+
+ * main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload
+ manager commands, which supplement the need to have Command
+ privilege. (closes issue #10736)
+
+ * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
+ main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for
+ FreeBSD... testing for every conceivable math function now
+
+2007-09-21 19:55 +0000 [r83500] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Fix compilation errors in CLI command
+ updates to SS7 CLI commands
+
+2007-09-21 19:54 +0000 [r83499] Matthew Fredrickson <creslin@digium.com>
+
+ * doc/ss7.txt (added): Add an SS7 readme for setup and use of
+ libss7 and asterisk
+
+2007-09-21 18:41 +0000 [r83484] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Fix some areas where we were still using '|'
+ for an argument delimiter (closes issue #10793)
+
+2007-09-21 18:27 +0000 [r83483] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Update app_queue to use commas as application
+ argument separators. (closes issue #10793, snar)
+
+2007-09-21 17:36 +0000 [r83466] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_manager.c: Fix cdr_manager, such that if the config file
+ is created past load, it'll start logging (and conversely, if the
+ config file is destroyed or deactivated, the logging is
+ disabled). Reported by Juggie via IRC, fix by me.
+
+2007-09-21 14:40 +0000 [r83433] Russell Bryant <russell@digium.com>
+
+ * res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c,
+ main/db1-ast/hash/hash.c, include/asterisk/channel.h,
+ channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c,
+ main/cdr.c, main/channel.c, channels/chan_misdn.c,
+ main/ast_expr2f.c, main/file.c, include/asterisk/sched.h,
+ channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c,
+ main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged
+ revisions 83432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) |
+ 4 lines gcc 4.2 has a new set of warnings dealing with cosnt
+ pointers. This set of changes gets all of Asterisk (minus
+ chan_alsa for now) to compile with gcc 4.2. (closes issue #10774,
+ patch from qwell) ........
+
+2007-09-21 14:25 +0000 [r83431] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
+ main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the
+ presence of trunc and round, and make the ISOC99 detection a
+ little more sane (closes issue #10776)
+
+2007-09-21 13:36 +0000 [r83401] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 83400 via svnmerge ........ r83400 | file |
+ 2007-09-21 10:34:32 -0300 (Fri, 21 Sep 2007) | 4 lines Fix video
+ under certain circumstances. It would have been possible for the
+ formats on the channel to not contain the video format. (closes
+ issue #10782) Reported by: cwhuang ........
+
+2007-09-20 23:14 +0000 [r83381] Jason Parker <jparker@digium.com>
+
+ * apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More
+ NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch
+ uploaded by moy (license 222) app_minivm.c.patch uploaded by
+ eliel (license 64) astmm.c.patch uploaded by eliel (license 64)
+
+2007-09-20 21:37 +0000 [r83350-83351] Mark Michelson <mmichelson@digium.com>
+
+ * /: Oops. Getting rid of svnmerge-integrated and automerge stuff
+
+ * /, apps/app_queue.c: Merging changes from queue_refcount_trunk
+ into trunk. Refcounted queues now in place.
+
+2007-09-20 21:17 +0000 [r83293-83349] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 83348 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) |
+ 4 lines When daemonizing, don't change working directory to "/".
+ It makes it not be able to do a core dump when not running as
+ uid=root. (closes issue #10766, xrg) ........
+
+ * /, contrib/scripts/safe_asterisk: Merged revisions 83316 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) |
+ 3 lines Change safe_asterisk to explicitly ask for /bin/bash, as
+ it uses bashisms. (closes issue #10772, reported by culrich)
+ ........
+
+ * main/dsp.c: trivial formatting change
+
+ * main/asterisk.c: trivial formatting change
+
+ * main/app.c: minor spelling fixes in a comment
+
+ * main/app.c: minor grammar fix
+
+ * channels/chan_sip.c: fix spelling in a comment
+
+ * main/asterisk.c: trivial formatting change
+
+2007-09-20 19:05 +0000 [r83251-83278] Jason Parker <jparker@digium.com>
+
+ * doc/modules.txt: Fix a trivial typo, to test our new commit bot
+
+ * /, apps/app_disa.c: Merged revisions 83246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8
+ lines If # is pressed after dialing an extension in DISA, stop
+ trying to collect more digits. (closes issue #10754) Reported by:
+ atis Patches: app_disa.c.branch.patch uploaded by atis (license
+ 242) app_disa.c.trunk.patch uploaded by atis (license 242)
+ ........
+
+2007-09-20 16:28 +0000 [r83234] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7
+ lines Make sure the minimum T1 timer value is obeyed in all
+ cases. (closes issue #10768) Reported by: flefoll Patches:
+ chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license
+ 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll
+ (license 244) ........
+
+2007-09-20 16:27 +0000 [r83233] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Don't start the event processing thread until
+ after forking. (reported by Simon on the -dev list, thanks!)
+
+2007-09-20 16:19 +0000 [r83229-83231] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7
+ lines Fix a minor spelling error. (closes issue #10769) Reported
+ by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded
+ by flefoll (license 244) chan_sip.c.br14.83070.inita-patch
+ uploaded by flefoll (license 244) ........
+
+ * pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks
+ in pbx_dundi, cdr_pgsql, and the configuration file parser.
+
+2007-09-19 23:16 +0000 [r83213] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c,
+ apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724)
+ Patches: chan_zap.c.patch uploaded by moy (license 222)
+ app_queue.c.patch uploaded by eliel (license 64)
+ app_voicemail.c.patch uploaded by eliel (license 64)
+ app_meetme.c.patch uploaded by eliel (license 64)
+
+2007-09-19 20:06 +0000 [r83182-83183] Joshua Colp <jcolp@digium.com>
+
+ * cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme
+ for me today?)
+
+ * res/res_adsi.c: Clean up code in res_adsi.
+
+2007-09-19 19:54 +0000 [r83176-83181] Russell Bryant <russell@digium.com>
+
+ * funcs/func_shell.c: put the channel in autoservice when executing
+ func_shell
+
+ * /, apps/app_system.c: Merged revisions 83179 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) |
+ 5 lines The System() and TrySystem() applications can take a
+ substantial amount of time to execute while not servicing the
+ channel. So, put the channel in autoservice while the command is
+ being executed. (closes issue #10726, reported by mnicholson)
+ ........
+
+ * funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) |
+ 4 lines Using curl can take a substantial amount of time, so the
+ channel should be autoserviced while waiting for it to complete.
+ (closes issue #10725, reported by mnicholson) ........
+
+ * /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) |
+ 8 lines When handling a reload of chan_iax2, don't use an
+ ao2_callback() to POKE all peers. Instead, use an iterator. By
+ using an iterator, the peers container is not locked while the
+ POKE is being done. It can cause a deadlock if the peers
+ container is locked because poking a peer will try to lock pvt
+ structs, while there is a lot of other code that will hold a pvt
+ lock when trying to go lock the peers container. (reported to me
+ directly by Loic Didelot. Thank you for the debug info!) ........
+
+2007-09-19 17:22 +0000 [r83155-83157] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_db.c: Fix indentation in app_db.
+
+ * apps/app_authenticate.c: Clean up code in app_authenticate.
+
+ * apps/app_adsiprog.c: Clean up code in app_adsiprog.
+
+2007-09-19 15:11 +0000 [r83126] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 83121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) |
+ 4 lines Fix up another potential race condition. Do the loop
+ decrementing use count on events with the eventq protected from
+ being changed. (reported on IRC by Ivan) ........
+
+2007-09-19 15:08 +0000 [r83105-83114] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_disa.c: DISA only needs to know about the end of DTMF,
+ not the beginning/duration.
+
+ * apps/app_disa.c: Clean up app_disa code a bit.
+
+2007-09-19 13:55 +0000 [r83076] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c: Replace Google namespace occurrences with
+ Jingle. The former namespace is handled by chan_gtalk.
+
+2007-09-19 13:49 +0000 [r83073-83075] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 83074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6
+ lines Protect the CDR record from modification by pbx_exec so
+ that the application data contains the Queue data. (closes issue
+ #10761) Reported by: snar Patches: app-queue-mixmonitor.patch
+ uploaded by snar (license 245) ........
+
+ * main/manager.c: Extend manager show connected with additional
+ information. (closes issue #10757) Reported by: outtolunc
+ Patches: manager.c.sessionstart.diff uploaded by outtolunc
+ (license 237)
+
+2007-09-19 13:29 +0000 [r83072] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c: Remove namespaces in payload-type tags.
+
+2007-09-19 13:21 +0000 [r83071] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6
+ lines (closes issue #10760) Reported by: dimas Patches:
+ chan_sip.patch uploaded by dimas (license 88) Read in
+ subscribecontext option in general to be the default. ........
+
+2007-09-19 12:23 +0000 [r83055] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c, include/asterisk/jingle.h: Transmit
+ proper invitation, thus conforming to XEP-0166 (Jingle general
+ specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176
+ (Jingle ICE Transport).
+
+2007-09-19 09:48 +0000 [r83025] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
+ channels/misdn_config.c: Merged revisions 83023-83024 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) |
+ 1 line added 'astdtmf' option to allow configuring the asterisk
+ dtmf detector instead of the mISDN_dsp ones. also added the patch
+ from irroot #10190, so that dtmf tones detected by the asterisk
+ detector are passed outofband to asterisk, to make any use of
+ dtmf tones at all. ........ r83024 | crichter | 2007-09-19
+ 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which
+ violates the coding guidelines. ........
+
+2007-09-19 00:21 +0000 [r82993] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_flash.c: Merged revisions 82992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) |
+ 4 lines Change the description of app_flash to note how it can be
+ a useful tool instead of just saying that it is generally a
+ worthless feature. (Thanks to Jim Van Meggelen for pointing it
+ out and providing the proposed text) ........
+
+2007-09-18 23:42 +0000 [r82962] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82961 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2
+ lines Initialize a variable to NULL to make the world happy.
+ ........
+
+2007-09-18 22:46 +0000 [r82931] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) |
+ 11 lines Add a new patch to handle interrupting the fgets() call
+ when using FastAGI. This version of the patch maintains the
+ original behavior of the code when not using FastAGI. (closes
+ issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch
+ uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch
+ uploaded by juggie (license 24) Slight mods by me Tested by:
+ juggie, festr ........
+
+2007-09-18 22:43 +0000 [r82871-82930] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c,
+ channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
+ res/res_musiconhold.c, res/res_jabber.c, main/manager.c,
+ res/res_agi.c, channels/chan_features.c, main/logger.c,
+ main/http.c, channels/chan_alsa.c, res/res_realtime.c,
+ res/res_odbc.c: (issue #10724) Reported by: eliel Patches:
+ res_features.c.patch uploaded by eliel (license 64)
+ res_agi.c.patch uploaded by seanbright (license 71)
+ res_musiconhold.c.patch uploaded by seanbright (license 71)
+ pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded
+ by moy (license 222) frame.c.patch uploaded by moy (license 222)
+ manager.c.patch uploaded by moy (license 222) http.c.patch
+ uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy
+ (license 222) res_realtime.c.patch uploaded by eliel (license 64)
+ res_odbc.c.patch uploaded by seanbright (license 71)
+ res_jabber.c.patch uploaded by eliel (license 64)
+ chan_local.c.patch uploaded by eliel (license 64)
+ chan_agent.c.patch uploaded by eliel (license 64)
+ chan_alsa.c.patch uploaded by eliel (license 64)
+ chan_features.c.patch uploaded by eliel (license 64)
+ chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch
+ (includes all of the above patches) uploaded by seanbright
+ (license 71) Convert many CLI commands to the NEW_CLI format.
+
+ * configs/voicemail.conf.sample, apps/app_voicemail.c: (closes
+ issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff
+ uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by
+ qwell (license 4) Tested by: callguy, ruffle Add an option to
+ disable the automatic moving of "heard" messages to the Old
+ folder.
+
+2007-09-18 20:59 +0000 [r82868] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 82867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) |
+ 10 lines Fix a memory leak that can occur on systems under higher
+ load. The issue is that when events are appended to the master
+ event queue, they use the number of active sessions as a use
+ count so it will know when all active sessions at the time the
+ event happened have consumed it. However, the handling of the
+ number of sessions was not properly synchronized, so the use
+ count was not always correct, causing an event to disappear
+ early, or get stuck in the event queue for forever. (closes issue
+ #9238, reported by bweschke, patch from Ivan, modified by me)
+ ........
+
+2007-09-18 20:10 +0000 [r82866] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep
+ 2007) | 4 lines Moving the logic for handling an empty membername
+ to the create_member function so that there is a common place
+ where this occurs instead of being spread out to several
+ different places. ........
+
+2007-09-18 19:06 +0000 [r82835] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82834 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007)
+ | 2 lines there is no need for conditional logic to select
+ ->interface or ->membername, snince ->membername will always be
+ populated ........
+
+2007-09-18 16:34 +0000 [r82803] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) |
+ 4 lines When copying the contents from the wildcard peer, do a
+ deep copy instead of shallow copy so that it doesn't crash when
+ beging destroyed. (closes issue #10546, patch by me) ........
+
+2007-09-18 16:16 +0000 [r82800] Jason Parker <jparker@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: (closes issue
+ #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch
+ uploaded by snar (license 245) queues.conf.patch uploaded by snar
+ (license 245) Add an updatecdr option to queues.conf, so that if
+ a "member name" is specified, the cdr record will be updated with
+ that, rather than the channel.
+
+2007-09-18 16:14 +0000 [r82776-82793] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/threadstorage.h: Make sure that libpthread
+ doesn't try to call free() directly when MALLOC_DEBUG is enabled.
+ If it does, Asterisk will crash as the address isn't the real
+ beginning of the allocation.
+
+ * channels/chan_zap.c: Don't use ast_channel_lock_both() here, it
+ only exists in one of my branches. This is theoretically a
+ potential deadlock, but it's the way it was before so I'm going
+ to leave it this way for now.
+
+2007-09-18 15:29 +0000 [r82752] Jason Parker <jparker@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500
+ (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains
+ option in SIP sample config. Issue 10753 ........
+
+2007-09-17 22:59 +0000 [r82728] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c,
+ channels/chan_agent.c, channels/chan_alsa.c,
+ channels/chan_iax2.c, channels/chan_mgcp.c: convert various
+ places that access the channel lock directly to use the channel
+ lock wrappers
+
+2007-09-17 21:52 +0000 [r82710-82712] Jason Parker <jparker@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Don't try to continue loading
+ cdr_sqlite3_custom on a module load failure (such as the config
+ not existing) Closes issue #10749, patch by seanbright.
+
+ * configs/http.conf.sample: Fix the sample redirect to point to a
+ valid file in the Asterisk GUI. Closes issue #10748, patch by
+ bkruse
+
+2007-09-17 20:24 +0000 [r82595-82679] Russell Bryant <russell@digium.com>
+
+ * doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support
+ for #include, var_metric, and cat_metric in res_config_sqlite
+ (closes issue #10738, rbraun_proformatique)
+
+ * /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged
+ revisions 82676 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) |
+ 4 lines Put a memset in ast_localtime() instead of a couple
+ places in app_voicemail to prevent the problem everywhere instead
+ of just a couple of places. (related to issue #10746) ........
+
+ * /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) |
+ 6 lines Initialize some memory to fix crashes when leaving
+ voicemail. This problem was fixed by running Asterisk under
+ valgrind. (closes issue #10746, reported by arcivanov, patched by
+ me) *** IMPORTANT NOTE: We need to check to see if this same bug
+ exists elsewhere. ........
+
+ * apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h,
+ apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
+ res/res_features.c, apps/app_queue.c, channels/chan_iax2.c,
+ pbx/pbx_config.c: Make the MALLOC_DEBUG output for free() useful
+ again. After changing calls to free to be ast_free, astmm said
+ all calls to free were coming from utils.h
+
+ * /, res/res_features.c: Merged revisions 82594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) |
+ 5 lines Handle the case where there are multiple dynamic features
+ with the same digit mapping, but won't always match the activated
+ on/by access controls. In that case, the code needs to keep
+ trying features for a match. (reported by Atis on the
+ asterisk-dev list, patched by me) ........
+
+2007-09-17 16:44 +0000 [r82593] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17
+ Sep 2007) | 2 lines fix a couple of places where a logical member
+ name (if specified) was not used, but instead the direct
+ interface was listed ........ r82592 | kpfleming | 2007-09-17
+ 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that
+ wasn't supposed to be committed... doh! ........
+
+2007-09-17 14:58 +0000 [r82568] Doug Bailey <dbailey@digium.com>
+
+ * main/http.c: Fix memory leak introduced when POST support was
+ added.
+
+2007-09-17 02:20 +0000 [r82516-82546] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: (closes issue #10715) Reported by:
+ the-chopper Don't bother hanging up the new channel if it does
+ not exist yet.
+
+ * main/pbx.c, /: Merged revisions 82514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4
+ lines (closes issue #10734) Reported by: asgaroth Instead of
+ passing a NULL pointer into snprintf pass "". It makes Solaris
+ much happier. ........
+
+2007-09-16 15:32 +0000 [r82496] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Option maxmessage should be maxsecs
+ per-folder, too (closes issue #10729)
+
+2007-09-14 21:30 +0000 [r82457] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 82444 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1
+ line closes issue #10668; thanks to arkadia for his patch; had to
+ leave out the bit about ending the previous cdr in the fork; it
+ would destroy current implementations. ........
+
+2007-09-14 21:21 +0000 [r82454] Russell Bryant <russell@digium.com>
+
+ * /, configs/zapata.conf.sample: Merged revisions 82435 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) |
+ 3 lines Add a note to help clarify the value set with the
+ echocancel option. (inspired by Malcolm's blog post on
+ blogs.digium.com about HPEC) ........
+
+2007-09-14 19:49 +0000 [r82401] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Add support
+ in chan_skinny for sending RTP directly to the endpoints. Closes
+ issue #9154, patch by DEA
+
+2007-09-14 18:37 +0000 [r82397-82400] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocking revision 82398
+
+ * /, apps/app_queue.c: Merged revisions 82396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep
+ 2007) | 5 lines Adding member name field to manager events where
+ they were missing before (closes issue #10721, reported by snar)
+ ........
+
+2007-09-14 17:51 +0000 [r82395] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5
+ lines If a channel does not have an owner, do not try to set a
+ channel variable. This will end up making the channel variable
+ global, which is not right. Closes issue #10720, patch by
+ flefoll. ........
+
+2007-09-14 17:29 +0000 [r82393] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute
+ method to res_odbc (closes issue #10722)
+
+2007-09-14 16:02 +0000 [r82386-82391] Russell Bryant <russell@digium.com>
+
+ * channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed),
+ channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c,
+ channels/xpmr/xpmr_coef.h: use the standard license header for
+ the xpmr files
+
+ * channels/chan_usbradio.c (added), channels/xpmr (added): Add
+ chan_usbradio to trunk
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ Merged revisions 82385 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) |
+ 3 lines Add checking for libusb here, so nobody has to deal with
+ conflicts in the chan_usbradio-1.4 branch every time the
+ configure script gets changed ........
+
+2007-09-14 14:44 +0000 [r82377] Mark Michelson <mmichelson@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14
+ Sep 2007) | 5 lines Fixing a typo in the coding guidelines
+ (closes issue #10717, reported and patched by leedm777) ........
+
+2007-09-14 13:02 +0000 [r82373] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c: Fix DTMF following what has been done in
+ issue #9401. Thanks irroot.
+
+2007-09-13 23:12 +0000 [r82359] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4
+ lines Fix a small typo. retrytime > waittime ........
+
+2007-09-13 21:53 +0000 [r82347-82352] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Changed "in" to "queue" in "queue
+ {pause|unpause} member" command to be more clear. Also added
+ check to be sure that sixth argument is the word "reason" if full
+ command is given
+
+ * CHANGES, apps/app_queue.c: Added the ability to pause and unpause
+ members via the CLI
+
+ * /, apps/app_queue.c: Merged revisions 82346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep
+ 2007) | 4 lines Preemptively fixing a possible segfault. It is
+ possible that queuename is NULL (meaning pause ALL queues), so
+ use q->name instead. ........
+
+2007-09-13 20:13 +0000 [r82345] Jason Parker <jparker@digium.com>
+
+ * /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9
+ lines Fix a crash that could occur in cdr_csv when mutliple
+ threads tried to close the same file. Do we actually need the
+ locking here? What happens if you open the same file twice, and
+ two threads try to write to it at the same time? Is fputs() going
+ to write out the entire line at once? I suspect that it could be
+ possible for the second fopen to run during the first fputs, so
+ the position could be in the middle of the previously written
+ line... Issue 10347, initial patch by explidous (but I removed
+ all of the paranoia stuff..) ........
+
+2007-09-13 19:16 +0000 [r82338-82341] Russell Bryant <russell@digium.com>
+
+ * /, main/astobj2.c: Merged revisions 82339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) |
+ 1 line resolve a warning when not building under dev mode
+ ........
+
+ * include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged
+ revisions 82337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) |
+ 4 lines Only compile in tracking astobj2 statistics if dev-mode
+ is enabled. Also, when dev mode is enabled, register the CLI
+ command that can be used to run the astobj2 test and print out
+ statistics. ........
+
+2007-09-13 18:13 +0000 [r82336] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, LICENSE: Merged revisions 82335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700
+ (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13
+ Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license
+ exceptions ........ ................
+
+2007-09-13 16:58 +0000 [r82329] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
+ setvar support to chan_zap. Just like you can in chan_sip and
+ chan_iax2 you can now use it with zaptel channels. (done while in
+ Montreal at the Asterisk bootcamp!)
+
+2007-09-13 16:27 +0000 [r82327] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82326 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep
+ 2007) | 7 lines Added logic to handle the unlikely case that
+ someone has two queues with the same name. Asterisk will log a
+ warning message letting the user know that one was already
+ defined with that name and is it skipping all further instances.
+ This also will work for realtime queues but in order for that to
+ happen, the user would have to trigger a perfectly timed reload
+ as a realtime queue is being looked up, which is highly unlikely
+ (but taken care of nonetheless). ........
+
+2007-09-13 15:26 +0000 [r82321] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h, doc/res_config_sqlite.txt,
+ res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various
+ code and documentation cleanups for res_config_sqlite (closes
+ issue #10711, rbraun_proformatique)
+
+2007-09-13 15:25 +0000 [r82312-82320] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c: Modify rule filters to match with the
+ Jingle namespace constant
+
+ * include/asterisk/jingle.h: Assign namespace properly
+
+ * channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle
+ and Jingle DTMF namespaces. As both specifications are in the
+ Experimental status, the namespaces specified therein shall be of
+ the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See
+ the Namespace issuance section in XEP-0053 :
+ http://www.xmpp.org/extensions/xep-0053.html#namespaces
+
+ * channels/chan_jingle.c: Reflect Jingle DTMF specification changes
+
+2007-09-13 13:34 +0000 [r82311] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Fix a missing unref of a member struct. This
+ was pointed out by Marta. Thanks! This function in 1.4 didn't
+ have the problem.
+
+2007-09-13 11:54 +0000 [r82310] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13
+ Sep 2007) | 4 lines Closes issue #9401, reported and patched by
+ irrot, with slight modifications by me. Handle DTMF sent by
+ Asterisk properly. ........
+
+2007-09-12 21:57 +0000 [r82297] Russell Bryant <russell@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 82296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) |
+ 3 lines Fix a check of the wrong pointer, as pointed out by an
+ XXX comment left in the code. The problem was harmless, however.
+ ........
+
+2007-09-12 21:55 +0000 [r82294] Jason Parker <jparker@digium.com>
+
+ * channels/chan_iax2.c: After some discussions, we decided that the
+ return values here were a bit messy. This also fixes a bug on
+ reload, where peers may not have reregistered properly.
+
+2007-09-12 21:32 +0000 [r82290-82292] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12
+ Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files
+ ........
+
+ * main/stdtime/private.h, /, main/stdtime/tzfile.h,
+ funcs/func_strings.c, apps/app_sms.c,
+ include/asterisk/localtime.h, main/stdtime/localtime.c: Merged
+ revisions 82285 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007)
+ | 4 lines Working on issue #10531 exposed a rather nasty 64-bit
+ issue on ast_mktime, so we updated the localtime.c file from
+ source. Next we'll have to write ast_strptime to match. ........
+
+2007-09-12 21:17 +0000 [r82289] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem
+ because unless get_member_status returned QUEUE_NORMAL, a NULL
+ member would be unreferenced. While this didn't cause any crashes
+ or anything terrible, it still is incorrect
+
+2007-09-12 20:50 +0000 [r82288] Steve Murphy <murf@digium.com>
+
+ * main/config.c: This fix closes issue #10642 -- it's not perfect,
+ but should retain most blank lines in config files, via
+ read/write cycles.
+
+2007-09-12 20:47 +0000 [r82287] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007)
+ | 1 line remove a race condition for the creation of
+ recordthread's, and fix a small memory leak. This closes issue#
+ 10636 ........
+
+2007-09-12 16:24 +0000 [r82283] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build
+ warnings (closes issue #10698, reported and patched by snuffy)
+
+2007-09-12 15:53 +0000 [r82279-82282] Russell Bryant <russell@digium.com>
+
+ * utils/hashtest2.c: Change the traversal to use ao2_callback()
+ instead of an ao2_iterator. Using ao2_callback() is a much more
+ efficient way of performing an operation on every item in the
+ container. This change makes hashtest2 run in about 25% of the
+ time it ran before on my system. In general, I would say that it
+ makes the most sense to use an ao2_iterator if the operation
+ being performed is going to take a long time and you don't want
+ to keep the container locked while you work with each object.
+ Otherwise, the use of ao2_callback is preferred.
+
+ * /, main/asterisk.c: Merged revisions 82280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) |
+ 4 lines Clean up the output of "asterisk -h". This tweaks the
+ wording and wraps lines at 80 characters. (closes issue #10699,
+ seanbright) ........
+
+ * /, res/res_agi.c: Merged revisions 82278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) |
+ 3 lines revert patch from issue #10553, as someone not using
+ fastagi reported that this broke their system. ........
+
+2007-09-12 14:31 +0000 [r82275-82277] Mark Michelson <mmichelson@digium.com>
+
+ * /: Blocking changes from revision 82276
+
+ * /, apps/app_queue.c: Merged revisions 82274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep
+ 2007) | 6 lines We should only initialize a realtime queue when
+ it is allocated, not every time we access it. This prevents the
+ members ao2_container from being reallocated every time the queue
+ is accessed. I also removed a debug message I had accidentally
+ left in on a previous commit. ........
+
+2007-09-11 23:07 +0000 [r82273] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Fix to make sure we don't hangup a call when
+ getting a RLC without sending REL. Found making sure we are Q.784
+ (the SS7 test specification) compliant
+
+2007-09-11 22:38 +0000 [r82269-82270] Russell Bryant <russell@digium.com>
+
+ * main/config.c: remove unused functions that made this file not
+ build under dev mode
+
+ * /, apps/app_queue.c: Merged revisions 82267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) |
+ 3 lines Fix incorrect uses of ao2_find(). Every one of these
+ calls was reading bogus memory ... ........
+
+2007-09-11 22:37 +0000 [r82268] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile, main/config.c: This solves an unreported solaris
+ compile problem (missing -lnsl -lsocket).
+
+2007-09-11 21:43 +0000 [r82266] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged
+ revisions 82265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4
+ lines (closes issue #10679) Reported by: andrew Build under dev
+ mode when K6OPTS is enabled. ........
+
+2007-09-11 20:50 +0000 [r82264] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) |
+ 5 lines Fix another missing unref of member objects. This one was
+ pointed out by Marta. When building the outgoing list in
+ try_calling(), a member reference is stored in each outgoing
+ entry. However, when this list got destroyed, the reference was
+ not released. ........
+
+2007-09-11 20:49 +0000 [r82262] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 82261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1
+ line this change should fix issue # 10659 -- what I worry about
+ is how many other bug reports it may generate. Hopefully, we can
+ please the/a majority. Hopefully. We shall see. Calls not marked
+ ANSWERED and with only one channel name will not be posted. This
+ should eliminate the double CDR's. ........
+
+2007-09-11 18:37 +0000 [r82257-82258] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample: Lil' bit more documentation to keep
+ folks happy.
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes
+ issue #9433) Reported by: junky Patches: register_trying.diff.txt
+ uploaded by jcmoore Disable sending 100 Trying on REGISTER
+ attempts and make it an option. This has been signed off by oej.
+
+2007-09-11 17:16 +0000 [r82256] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile: fixing up the pthread stuff for hashtest2
+
+2007-09-11 16:15 +0000 [r82254] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged
+ revisions 82249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) |
+ 1 line fixed a hold/retrieve issue. ........
+
+2007-09-11 16:12 +0000 [r82253] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep
+ 2007) | 6 lines All instances of ao2_iterators which were just
+ named 'i' have been renamed to 'mem_iter' so that when refcounted
+ queues are merged into trunk, there will be little confusion
+ regarding iterator names, especially when a queue and member
+ iterator are used in the same function. ........
+
+2007-09-11 16:05 +0000 [r82251] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) |
+ 4 lines The sample dundi.conf claims support for a wildcard peer
+ entry - [*], but the code did not support it. This patch makes it
+ work. (closes issue #10546, patch by dds, with some changes by
+ me) ........
+
+2007-09-11 15:34 +0000 [r82248] Joshua Colp <jcolp@digium.com>
+
+ * main/cdr.c: (closes issue #10666) Reported by: arkadia Patches:
+ cdr_lockorder.patch uploaded by arkadia (license 233) Optimize
+ CDR stuff a bit.
+
+2007-09-11 15:31 +0000 [r82246-82247] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: Remove an unused variable. I have no idea why this
+ was marked with the unused attribute instead of just removing it.
+ :)
+
+ * /, res/res_agi.c: Merged revisions 82245 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) |
+ 9 lines (closes issue #10553) Reported by: juggie Patches:
+ res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
+ juggie When using fastagi, fgets() can return before a full line
+ is read. Add explicit handling for the case where it gets
+ interrupted. ........
+
+2007-09-11 14:58 +0000 [r82242-82244] Joshua Colp <jcolp@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6
+ lines (closes issue #10577) Reported by: jamesgolovich Patches:
+ asterisk-dundifree.diff.txt uploaded by jamesgolovich (license
+ 176) Don't leak memory when unloading DUNDi. ........
+
+ * apps/app_meetme.c: (closes issue #10560) Reported by: ruffle
+ Patches: rb uploaded by ruffle (license 201) Show whether the
+ conference is locked or not on the CLI.
+
+2007-09-11 14:35 +0000 [r82237-82241] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82240 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) |
+ 2 lines Add a couple more missing unrefs of queue member objects
+ ........
+
+ * /, apps/app_queue.c: Merged revisions 82238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) |
+ 2 lines Add a missing unref of a queue member in an error
+ handling block ........
+
+ * /, apps/app_queue.c: Merged revisions 82236 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) |
+ 2 lines Document why membercount can not simply be replaced by
+ ao2_container_count() ........
+
+2007-09-11 13:46 +0000 [r82231-82235] Joshua Colp <jcolp@digium.com>
+
+ * utils/Makefile: Include string compatibility file in hashtest2.
+
+ * utils/hashtest2.c: Include compat.h to hopefully make it
+ compatible with FreeBSD.
+
+ * utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h
+ exists before including it.
+
+ * main/manager.c: (closes issue #10695) Reported by: junky Patches:
+ count_showconn.diff uploaded by junky (license 177) Provide a
+ count of connected users to manager.
+
+ * main/minimime/minimime.c, main/minimime/tests/create.c,
+ main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes
+ issue #10692) Reported by: snuffy Patches: minivm.diff uploaded
+ by snuffy (license 35) Instead of using err (which is not
+ available under Solaris) use fdprintf with stderr.
+
+2007-09-10 20:03 +0000 [r82200] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to
+ have manager-style output, instead of CLI-style output (closes
+ issue #8254)
+
+2007-09-10 19:56 +0000 [r82199] Russell Bryant <russell@digium.com>
+
+ * /: Blocked revisions 82198 via svnmerge ........ r82198 | russell
+ | 2007-09-10 14:53:17 -0500 (Mon, 10 Sep 2007) | 3 lines backport
+ astobj2 race condition fix. This function is the exact same as
+ trunk so it applies here as well. ........
+
+2007-09-10 19:10 +0000 [r82185] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fixing a problem where NULL channels would
+ cause a crash when calling indisposed queue members (i.e. paused,
+ wrapup time not completed, etc.)
+
+2007-09-10 18:32 +0000 [r82178] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 82155 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007)
+ | 2 lines Convert struct member to use refcounts (closes issue
+ #10199) ........
+
+2007-09-10 17:39 +0000 [r82154] Jason Parker <jparker@digium.com>
+
+ * main/db.c: Add a counter to the 'database deltree' CLI command.
+ Note: this is slightly different than the initial patch, because
+ I felt that using res <= 0 would be a change in behavior. Closes
+ issue #10687, patch by junky
+
+2007-09-10 16:59 +0000 [r82140] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile, utils/hashtest2.c (added): Committing my test for
+ astobj2, hashtest2.c, along with makefile changes in utils.
+
+2007-09-10 16:24 +0000 [r82125] Jason Parker <jparker@digium.com>
+
+ * main/db.c: Add counter to 'database show' CLI command. (also a
+ minor whitespace change that I found along the way) Closes issue
+ #10683, patch by junky
+
+2007-09-10 16:19 +0000 [r82124] Steve Murphy <murf@digium.com>
+
+ * main/astobj2.c: Changes applied from marta's team/marta/astobj2
+ branch to solve a race condition
+
+2007-09-10 15:05 +0000 [r82092] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10
+ Sep 2007) | 5 lines Removing non-existent options from misdn
+ configuration sample. (closes issue #10678, reported and patched
+ by IgorG) ........
+
+2007-09-10 14:26 +0000 [r82062-82077] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: (closes issue #10688) Reported by: casper
+ Patches: chan_sip.c.82076.diff uploaded by casper (license 55)
+ Remove double check for zombie flag and optimize things a bit.
+
+ * res/res_agi.c: (closes issue #10684) Reported by: junky Patches:
+ debug.diff uploaded by junky (license 177) Fix issue with debug
+ always showing up.
+
+ * apps/app_meetme.c: (closes issue #10686) Reported by: junky
+ Patches: meet.diff uploaded by junky (license 177) Change NOTICE
+ message to DEBUG.
+
+2007-09-09 02:45 +0000 [r82029] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08
+ Sep 2007) | 2 lines Fix inline compiles on really old compilers
+ (who uses gcc 2.7 anymore, really?) (closes issue #10675)
+ ........
+
+2007-09-08 19:01 +0000 [r81998-81999] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/slinfactory.h: Add doxygen documentation for
+ slinfactory_destroy(), mainly just noting that it doesn't free
+ the slinfactory itself. (This isn't related to a bug, i'm just
+ looking over random code)
+
+ * /, main/asterisk.c: Merged revisions 81997 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) |
+ 2 lines Fix a small memory leak. ast_unregister_atexit() did not
+ free the entry it removed. ........
+
+2007-09-08 16:37 +0000 [r81984] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Make Callerid more consistent in IMAP mail
+ headers (closes issue #10056, reported and patched by jaroth,
+ with small modification by me)
+
+2007-09-08 13:45 +0000 [r81953] Russell Bryant <russell@digium.com>
+
+ * /, .cleancount: Merged revisions 81952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) |
+ 11 lines (closes issue #10672) Bump the cleancount so that a
+ "make clean" will be forced. This is needed because my fix in
+ revision 81599 made a change to a data structure in file.h, and
+ since file dependency tracking is only on with dev-mode enabled,
+ file format modules that don't get rebuilt may crash, as is the
+ case with this issue. This makes me wonder - how much faster does
+ the code build without the file dependency tracking enabled? If
+ it doesn't make much of a difference, then it may be worth just
+ keeping it on all of the time, or perhaps just not in release
+ tarballs, so that this type of issue is avoided. ........
+
+2007-09-07 19:53 +0000 [r81910-81924] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500
+ (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable
+ to be used as the mixmonitor filename. This moves the setting of
+ the MEMBERINTERFACE variable to before mixmonitor. Issue 10671,
+ patch by sim. ........
+
+ * apps/app_queue.c: Add an optional reason parameter to
+ PauseQueueMember/UnpauseQueueMember applications and manager
+ events. Issue 8738, patch by rgollent
+
+2007-09-07 15:29 +0000 [r81891] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 81886 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep
+ 2007) | 3 lines Moving the explanation for joinempty to a more
+ appropriate place ........
+
+2007-09-07 12:32 +0000 [r81858-81873] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Don't check for epoll support when cross
+ compiling.
+
+ * main/channel.c, main/audiohook.c: Fix memory issue that crept up
+ with Russell's testing. It is *not* proper to free the frame we
+ get in ast_write.
+
+2007-09-06 22:32 +0000 [r81839-81849] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix the build ... oops
+
+ * /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) |
+ 16 lines (closes issue #9724, closes issue #10374) Reported by:
+ kenw Patches: 9724.txt uploaded by russell (license 2) Tested by:
+ kenw, russell Resolve a deadlock that occurs when doing a SIP
+ transfer to parking. I come across this type of deadlock fairly
+ often it seems. It is very important to mind the boundary between
+ the channel driver and the core in respect to the channel lock
+ and the channel-pvt lock. Channel drivers lock to lock the pvt
+ and then the channel once it calls into the core, while the core
+ will do it in the opposite order. The way this is avoided is by
+ having channel drivers either release their pvt lock while
+ calling into the core, or such as in this case, unlocking the pvt
+ just long enough to acquire the channel lock. ........
+
+2007-09-06 22:06 +0000 [r81827] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 81826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1
+ line We added COPTS for ASTCFLAGS additions, but not LDOPTS for
+ ASTLDFLAGS. This adds LDOPTS ........
+
+2007-09-06 21:01 +0000 [r81814] Joshua Colp <jcolp@digium.com>
+
+ * channels/iax2-parser.c: Initialize iax_frames variable to NULL,
+ keeps valgrind happy.
+
+2007-09-06 20:54 +0000 [r81783-81813] Russell Bryant <russell@digium.com>
+
+ * CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE()
+ function that can retrieve the state of an extension that has a
+ hint. (closes issue #10635, adamgundy)
+
+ * CHANGES: s/DEVSTATE/DEVICE_STATE/
+
+ * funcs/func_devstate.c: Rename the DEVSTATE() function to
+ DEVICE_STATE() to better conform to how other functions are
+ named. (inspired by issue #10635)
+
+ * CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function
+ from my sandbox branch into trunk. This function will let you
+ retrieve the list of devices or name associated with a hint.
+ (inspired by issue #10635)
+
+2007-09-06 20:16 +0000 [r81782] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_skinny.c, CHANGES: (closes issue #10377) Reported
+ by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak
+ (license 7) Add skinny show device, skinny show line, and skinny
+ show settings CLI commands.
+
+2007-09-06 20:05 +0000 [r81781] Russell Bryant <russell@digium.com>
+
+ * configs/extensions.conf.sample: Fix the syntax of declaring a
+ hint with a name to be compatible with trunk
+
+2007-09-06 20:00 +0000 [r81779] Jason Parker <jparker@digium.com>
+
+ * /, include/asterisk/astobj2.h: Merged revisions 81778 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2
+ lines This should fix a build issue that people building against
+ uClibc were seeing with the addition of astobj2 ........
+
+2007-09-06 19:43 +0000 [r81777] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7
+ lines (closes issue #10122) Reported by: stevefeinstein Patches:
+ meetme-unmute-manager.diff uploaded by qwell (license 4) Tested
+ by: stevefeinstein After looking over the code I agree with
+ Qwell. Setting the file descriptor to conference each time just
+ causes a fight back and forth. ........
+
+2007-09-06 17:00 +0000 [r81745] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged
+ revisions 81743 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007)
+ | 1 line Various string length fixes. Removed an unused variable
+ in aji_client structure (context) ........
+
+2007-09-06 16:57 +0000 [r81744] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/safe_asterisk: Incorporate the ability to log
+ output of safe_asterisk to syslog (closes issue #9882)
+
+2007-09-06 16:38 +0000 [r81742] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Patch on 10575. Add support for unequipped
+ CIC (UCIC) message as well as improve some of our CIC flags in
+ chan_zap
+
+2007-09-06 16:31 +0000 [r81730] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep
+ 2007) | 6 lines Fixes an issue where valid DTMF had to be pressed
+ twice to exit a queue if a member's phone was ringing. (closes
+ issue #10655, reported by strider2k, patched by me) ........
+
+2007-09-06 15:43 +0000 [r81712] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: various changes to
+ the documentation, and redefinition of ao2_hash_fn and
+ ao2_callback_fn typedefs, in preparation to more cleanup of the
+ _search_flags Please do not merge this change to 1.4 yet - there
+ are no functional changes anyways.
+
+2007-09-06 15:21 +0000 [r81683] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_features.c: Merged revisions 81682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep
+ 2007) | 5 lines Fixes a memory leak (closes issue #10658,
+ reported and patched by Ivan) ........
+
+2007-09-06 14:24 +0000 [r81651] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, res/res_jabber.c: Merged revisions 81650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007)
+ | 3 lines According to both RFC 3920 - section 9.1.2 - and
+ Google's XMPP server complaint, if set, the 'from' attribute must
+ be set to the user's full JID. ........
+
+2007-09-05 21:59 +0000 [r81632] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Not having this epoll specific code in
+ wait_for_answer was causing app_queue to infinitely loop. This
+ makes it so it doesn't. Thanks to file for pointing out where the
+ problem was and showing a similar function in app_dial as an
+ example of how to fix it.
+
+2007-09-05 21:45 +0000 [r81631] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05
+ Sep 2007) | 2 lines Solaris x86 compatibility fix ........
+
+2007-09-05 20:58 +0000 [r81601] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller
+
+2007-09-05 20:58 +0000 [r81600] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/file.h, /, main/say.c, res/res_features.c,
+ main/file.c, include/asterisk/channel.h: Merged revisions 81599
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) |
+ 11 lines Fix an issue that can occur when you do an attended
+ transfer to parking. If you complete the transfer before the
+ announcement of the parking spot finishes, then the channel being
+ parked will hear the remainder of the announcement. These changes
+ make it so that will not happen anymore. Basically, res_features
+ sets a flag on the channel is playing the announcement to so that
+ the file streaming core knows that it needs to watch out for a
+ channel masquerade, and if it occurs, to abort the announcement.
+ (closes BE-182) ........
+
+2007-09-05 16:48 +0000 [r81568] Tilghman Lesher <tlesher@digium.com>
+
+ * utils: Add two more generated files (requested by mvanbaak via
+ irc)
+
+2007-09-05 16:31 +0000 [r81560] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/devicestate.h, res/res_config_odbc.c,
+ channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c,
+ res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c,
+ include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c,
+ res/res_config_sqlite.c, include/asterisk/sha1.h,
+ include/asterisk/stringfields.h, include/asterisk/features.h:
+ Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy
+
+2007-09-05 15:32 +0000 [r81526-81535] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Weird. When I merged my changes from 1.4, they
+ merged into the wrong function. This should fix the build for
+ trunk.
+
+ * /, apps/app_queue.c: Merged revisions 81525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep
+ 2007) | 4 lines Fixing the build... ........
+
+2007-09-05 15:16 +0000 [r81524] Jason Parker <jparker@digium.com>
+
+ * channels/chan_phone.c, /: Merged revisions 81523 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500
+ (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL
+ channel tech. Also changed load_module function to use defines
+ rather than numbers for return values. Issue 10651, patch by
+ rbraun_proformatique, with additions by me. ........
+
+2007-09-05 15:04 +0000 [r81522] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81520 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep
+ 2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only
+ count members who are logged in and available. (related to issue
+ #10652, reported by wuwu) ........
+
+2007-09-05 14:47 +0000 [r81519] Steve Murphy <murf@digium.com>
+
+ * include/asterisk/config.h, main/config.c: this set of changes
+ fixes issue # 10643 by keeping track of the last object defined
+ in a file, and attaching any accumulated comments to that object
+ (category header or variable declaration). The file_save routine
+ also had to be upgraded to output these trailing comments.
+ Config.h was modified to include the trailing comment list on
+ categories and variables.
+
+2007-09-05 13:13 +0000 [r81459-81493] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 81492 via svnmerge ........ r81492 | file |
+ 2007-09-05 10:11:48 -0300 (Wed, 05 Sep 2007) | 4 lines (closes
+ issue #10650) Reported by: tacvbo Only print out that the spy was
+ removed while holding the spy lock. ........
+
+ * main/editline/sys.h: Finish up commit from revision 81452 by
+ removing last remnants of strlcat/strlcpy checks.
+
+2007-09-04 20:59 +0000 [r81454-81456] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 81455 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500
+ (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file,
+ we can just check whether it exists. Issue 10634, patch by me,
+ testing by pabelanger, sanity checked by bweschke ........
+
+ * /, configs/followme.conf.sample: Merged revisions 81453 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500
+ (Tue, 04 Sep 2007) | 4 lines Change default followme config file
+ to point to the correct files. Issue 10644, patch by pabelanger
+ ........
+
+2007-09-04 19:51 +0000 [r81445-81452] Russell Bryant <russell@digium.com>
+
+ * main/editline/configure, main/editline/configure.in: Don't check
+ for and include strlcpy and strlcat in editline. We also include
+ them directly in Asterisk. For platforms that need them (like my
+ mac), you will get a linker error due to the functions being
+ included twice.
+
+ * /, include/asterisk/astobj2.h, channels/chan_iax2.c,
+ main/astobj2.c: Merged revisions 81448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) |
+ 4 lines Remove the typedefs on ao2_container and ao2_iterator.
+ This is simply because we don't typedef objects anywhere else in
+ Asterisk, so we might as well make this follow the same
+ convention. ........
+
+ * include/asterisk/logger.h: logger.h depends on options.h, so go
+ ahead and include it
+
+2007-09-04 16:41 +0000 [r81443] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007)
+ | 2 lines there is no point in sending 401 Unauthorized to a UAS
+ that sent us a properly-formatted Authentication header with the
+ expected username and nonce but an incorrect response (which
+ indicates the shared secret does not match)... instead, let's
+ send 403 Forbidden so that the UAS doesn't retry with the same
+ authentication credentials repeatedly ........
+
+2007-09-04 14:28 +0000 [r81436-81441] Joshua Colp <jcolp@digium.com>
+
+ * configs/extensions.ael.sample: (closes issue #10633) Reported by:
+ pabelanger Patches: extensions.ael.sample.patch uploaded by
+ pabelanger (license 224) Update extensions.ael.sample with
+ voicemail and | changes.
+
+ * /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6
+ lines (closes issue #10632) Reported by: jamesgolovich Patches:
+ asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich
+ (license 176) Fix memory leak when unloading chan_iax2. The
+ firmware files were not being freed. ........
+
+ * main/channel.c, /: Merged revisions 81437 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4
+ lines (closes issue #10476) Reported by: mdu113 Only look for the
+ end of a digit when waiting for a digit. This in turn disables
+ emulation in the core. ........
+
+ * /, main/dns.c: Merged revisions 81435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7
+ lines (closes issue #10610) Reported by: john Patches:
+ dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
+ Don't return a match if no SRV record actually exists. ........
+
+2007-09-03 18:59 +0000 [r81434] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) |
+ 5 lines Remove a couple of calls to ast_string_field_free_pools()
+ on peers in error handling blocks in the code for building peers.
+ The peer object destructor does this and doing it twice will
+ cause a crash. (closes issue #10625, reported by and patched by
+ pnlarsson) ........
+
+2007-09-03 18:01 +0000 [r81430-81432] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c: Once we get past the file checks, we're loading,
+ so clear the FILEUNCHANGED flag (fixes #include) (closes issue
+ #10629)
+
+ * /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007)
+ | 2 lines The IF() function was not allowing true values that had
+ embedded colons (closes issue #10613) ........
+
+ * main/config.c: We shouldn't use a filename blindly without
+ checking to make sure it's unused first
+
+2007-09-01 06:03 +0000 [r81427] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
+ 81426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep
+ 2007) | 4 lines Making match_by_addr into ao2_match_by_addr and
+ making it available everywhere since it could be a handy callback
+ to have ........
+
+2007-08-31 21:29 +0000 [r81419] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/astobj2.h: Merged revisions 81418 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) |
+ 2 lines Remove references to a debugging parameter that does not
+ exist ........
+
+2007-08-31 19:50 +0000 [r81417] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug
+ 2007) | 6 lines Fixed broken behavior of a reload on realtime
+ queues. Prior to this patch, if a reload was issued and a
+ realtime queue had callers waiting in it, then the queue would be
+ removed from the queue list, but it would not actually be freed
+ (in fact, a debug message warning about a memory leak would come
+ up). With this patch, reloads do not touch realtime queues at
+ all. ........
+
+2007-08-31 18:46 +0000 [r81413] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 81412 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500
+ (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line
+ with the existing alpha order. Issue 10621, initial patch by
+ junky ........
+
+2007-08-31 17:43 +0000 [r81411] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31
+ Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command
+ available. Closes issue 10548, reported by keepitcool. ........
+
+2007-08-31 15:58 +0000 [r81408] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007)
+ | 2 lines add missing "transcoder show" (and deprecated "show
+ transcoder") CLI commands that were in 1.2 but never added to 1.4
+ ........
+
+2007-08-31 15:54 +0000 [r81402-81407] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_speech.c: Merged revisions 81406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2
+ lines Make it the engine's responsible to check for the presence
+ of results. ........
+
+ * /, res/res_features.c: Merged revisions 81403 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4
+ lines (closes issue #10618) Reported by: dimas Don't pass through
+ the stopped sounds frame.... just drop it. ........
+
+ * /, res/res_features.c: Merged revisions 81401 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4
+ lines (closes issue #10009) Reported by: dimas Don't output a
+ bridge failed warning message if it failed because one of the
+ channels was part of the masquerade process. That is perfectly
+ normal. ........
+
+2007-08-30 23:52 +0000 [r81400] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: Add new queryable fields from zaptel to 'zap
+ show status'
+
+2007-08-30 22:08 +0000 [r81398] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug
+ 2007) | 7 lines Removing an extraneous (and possibly misleading)
+ log message. Firstly, if the announce file isn't found, the
+ streaming functions will report it. Secondly, not all non-zero
+ returns from play_file mean that the announce file wasn't found.
+ Positive return values simply mean that a digit was pressed (most
+ likely to skip through the announcement). (closes issue #10612,
+ reported and patched by dimas) ........
+
+2007-08-30 21:25 +0000 [r81394-81396] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6
+ lines (closes issue #10514) Reported by: casper Patches:
+ chan_sip.c.80129.diff uploaded by casper (license 55) Remove
+ needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it
+ to ever be that value. ........
+
+ * channels/chan_sip.c: (closes issue #10565) Reported by: tootai
+ Make sure the external IP address has the standard SIP port set
+ for when the user does not specify the port in the externip
+ setting.
+
+2007-08-30 21:16 +0000 [r81393] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 81392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1
+ line via issue 10599, where 'CDR already initialized' messages
+ are being generated. Since all channels will have an init'd CDR
+ attached at creation time, this message is now particularly
+ useless. Removed. ........
+
+2007-08-30 20:55 +0000 [r81391] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_minivm.c: (closes issue #10336) Reported by: junky
+ Patches: minivm_output2.diff uploaded by junky (license 177)
+ Change console output of minivm show stats to be more simple for
+ external parsing.
+
+2007-08-30 20:31 +0000 [r81389-81390] Tilghman Lesher <tlesher@digium.com>
+
+ * main/sched.c: A schedule id of 0 is not possible and is used to
+ flag that we want to add a new item
+
+ * apps/app_readexten.c: Change wording as requested by Kevin
+
+2007-08-30 18:52 +0000 [r81388] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample: Added note to sample queues.conf file
+ to line up with most recent change regarding setinterfacevar.
+ MEMBERREALTIME indicates whether a member is realtime.
+
+2007-08-30 17:51 +0000 [r81387] Tilghman Lesher <tlesher@digium.com>
+
+ * main/logger.c: Always force reread of the config when we're
+ rotating the log file (closes issue #10598)
+
+2007-08-30 15:40 +0000 [r81384] Russell Bryant <russell@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 81383 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) |
+ 3 lines Add missing checks for the PTRACING define. (closes issue
+ #10559, paravoid) ........
+
+2007-08-30 15:36 +0000 [r81382] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug
+ 2007) | 3 lines Changed some manager event messages to reflect
+ whether a queue member is a realtime member or not ........
+
+2007-08-30 15:34 +0000 [r81380] Russell Bryant <russell@digium.com>
+
+ * configs/modem.conf.sample (removed), /, configs/enum.conf.sample,
+ configs/extensions.ael.sample: Merged revisions 81379 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) |
+ 3 lines Fix a typo, update a reload command, and remove an unused
+ configuration file. (closes issue #10606, casper) ........
+
+2007-08-30 15:24 +0000 [r81378] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN
+ function (closes issue #10082)
+
+2007-08-30 14:55 +0000 [r81377] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 81375 via svnmerge ........ r81375 | file |
+ 2007-08-30 11:53:43 -0300 (Thu, 30 Aug 2007) | 6 lines (closes
+ issue #10603) Reported by: jmls Patches: pbx.diff uploaded by
+ jmls (license 141) Backport changes from 81372. Add REASON
+ dialplan variable for when an originated call fails and the
+ failed extension is executed. ........
+
+2007-08-30 14:54 +0000 [r81376] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30
+ Aug 2007) | 1 line Fixed some warnings. ........
+
+2007-08-30 14:42 +0000 [r81370-81372] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls
+ Patches: pbx.diff uploaded by jmls (license 141) Add REASON
+ dialplan variable for when an originated call fails and the
+ failed extension is executed.
+
+ * /, res/res_features.c: Merged revisions 81369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4
+ lines (issue #10599) Reported by: dimas Handle the -1 control
+ subclass during feature dialing (it indicates to stop sounds).
+ ........
+
+2007-08-30 08:50 +0000 [r81368] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
+ revisions 81367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) |
+ 11 lines Fixed a severe issue where a misdn_read would lock the
+ channel, but read would not return because it blocks. later
+ chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER
+ which could result in a deadlock situation. misdn_read will now
+ not block forever anymore, and we don't queue the ANSWER frame at
+ all when we already was called with misdn_answer -> answer would
+ be called twice. Also we don't explicitly send a RELEASE_COMPLETE
+ on receiption of a RELEASE anymore, because mISDN does that for
+ us, this resulted in a problem on some switches, which would
+ block our port after some calls for a short while. ........
+
+2007-08-29 22:05 +0000 [r81365] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Added the MEMBERREALTIME variable when using
+ setinterfacevar in queues.conf
+
+2007-08-29 21:55 +0000 [r81364] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/event.h: Make the event header file work under
+ C++.
+
+2007-08-29 21:30 +0000 [r81363] Steve Murphy <murf@digium.com>
+
+ * main/config.c: init newer so compile won't complain.
+
+2007-08-29 21:25 +0000 [r81362] Russell Bryant <russell@digium.com>
+
+ * main/config.c: make trunk build again. murf will have to review
+ this to see if it was the right fix, as it is related to his last
+ change.
+
+2007-08-29 20:55 +0000 [r81361] Steve Murphy <murf@digium.com>
+
+ * res/res_config_pgsql.c, channels/chan_sip.c,
+ include/asterisk/config.h, channels/chan_iax2.c,
+ channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c,
+ main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c,
+ main/manager.c, channels/chan_skinny.c, apps/app_minivm.c,
+ main/http.c, utils/extconf.c, apps/app_directory.c,
+ apps/app_parkandannounce.c, apps/app_voicemail.c: This code was
+ in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I
+ didn't add it to 1.4 yet, because it's not entirely clear to me
+ if this is a bug fix or an enhancement. A lot of files were
+ affected by small changes like ast_variable_new getting an added
+ arg, for the file name the var was defined in; ast_category_new
+ gets added args of filename and lineno; ast_category and
+ ast_variable structures now record file and lineno for each
+ entry; a list of all #include and #execs in a config file (or any
+ of its inclusions are now kept in the ast_config struct; at save
+ time, each entry is put back into its proper file of origin, in
+ order. #include and #exec directives are folded in properly.
+ Headers indicating that the file was generated, are generated
+ also for each included file. Some changes to main/manager.c to
+ take care of file renaming, via the UpdateConfig command.
+ Multiple inclusions of the same file are handled by exploding
+ these into multiple include files, uniquely named. There's
+ probably more, but I can't remember it right now.
+
+2007-08-29 19:41 +0000 [r81353-81356] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Try to clarify the rules on changing ast_event and
+ ast_event_ie
+
+ * main/event.c: Fix parenthesis from my last commit
+
+ * main/event.c: Change pointer aritmetic on void * to char *
+
+ * main/event.c: there is not actually code that sends these over
+ the network in trunk yet
+
+2007-08-29 16:39 +0000 [r81350] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug
+ 2007) | 12 lines This patch, in essence, will correctly pause a
+ realtime queue member and reflect those changes in the realtime
+ engine. (issue #10424, reported by irroot, patch by me) This
+ patch creates a new function called update_realtime_member_field,
+ which is a generic function which will allow any one field of a
+ realtime queue member to be updated. This patch only uses this
+ function to update the paused status of a queue member, but it
+ lays the foundation for persisting the state of a realtime member
+ the same way that static members' state is maintained when using
+ the persistentmembers setting ........
+
+2007-08-29 16:25 +0000 [r81348] Joshua Colp <jcolp@digium.com>
+
+ * main/event.c: Return ast_event_get_ie_raw to using an iterator
+ and fix logic in ast_event_iterator_next.
+
+2007-08-29 16:09 +0000 [r81347] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug
+ 2007) | 3 lines Changed some tabs to spaces ........
+
+2007-08-29 16:07 +0000 [r81344-81345] Joshua Colp <jcolp@digium.com>
+
+ * main/event.c: This concludes bringing trunk back to a working
+ state.
+
+ * include/asterisk/event.h, main/event.c: To keep others happy...
+ revert part of my additions so trunk works.
+
+2007-08-29 15:59 +0000 [r81343] Russell Bryant <russell@digium.com>
+
+ * /, main/Makefile: Merged revisions 81342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) |
+ 3 lines If chan_h323 is not being built, don't use g++ to do the
+ final link of Asterisk. (in response to a question on the
+ asterisk-dev list) ........
+
+2007-08-29 15:57 +0000 [r81341] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug
+ 2007) | 8 lines This fix creates a more accurate way of detecting
+ whether realtime members were deleted. (closes issue 10541,
+ reported by Alric, patched by me) The REALLY nice things about
+ this patch is that queue members now have a "realtime" field
+ which will be true if the member is a realtime member. This means
+ we can check this value prior to certain processing if it should
+ ONLY be done for realtime members. ........
+
+2007-08-29 15:21 +0000 [r81335] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Changed one too many variable settings in
+ issue #9315 (closes issue #10592)
+
+2007-08-29 15:19 +0000 [r81334] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/event.h, include/asterisk/event_defs.h,
+ main/event.c: Add API calls for iterating through an event. This
+ should allow events to have multiple information elements (while
+ there was nothing preventing it before you could not actually
+ access any except the first one).
+
+2007-08-29 14:19 +0000 [r81333] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue
+ #10591, reported and patched by junky, with small modification by
+ me)
+
+2007-08-29 14:16 +0000 [r81326-81332] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4
+ lines (closes issue #9690) Reported by: mattv Make rtp timeouts
+ work even if two RTP streams are directly bridged in the RTP
+ stack. ........
+
+ * include/asterisk/utils.h: Add inline function for signed linear
+ subtraction.
+
+2007-08-28 21:39 +0000 [r81292] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) |
+ 3 lines Change the message about receiving a mini-frame before
+ the first full voice frame to a DEBUG message. ........
+
+2007-08-28 21:35 +0000 [r81290] Joshua Colp <jcolp@digium.com>
+
+ * main/logger.c: Add some read/write locking magic to make logger
+ reload operate again.
+
+2007-08-28 20:03 +0000 [r81277] Tilghman Lesher <tlesher@digium.com>
+
+ * main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support
+ better rotation of log files to be more like system logging
+ (closes issue #10398)
+
+2007-08-28 19:12 +0000 [r81227-81264] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/audiohook.h: Change the audiohook lock and
+ unlock wrappers to macros instead of inline functions. As inline
+ functions, the lock debug information will show that these are
+ always locked in audiohooks.h instead of the file where the lock
+ was actually acquired.
+
+ * funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking
+ around the uses of datastore_add and _find. There are still more
+ places in the tree that I have not yet changed if someone wants
+ to go through and find the places they are used without the
+ channel locked.
+
+ * main/channel.c, funcs/func_volume.c, include/asterisk/channel.h:
+ * Constify the uid field of channel datastores * Convert some
+ spaces to tabs in func_volume * Add a note in channel.h making it
+ clear that none of the datastore API calls lock the channel they
+ are given, so the channel should be locked before calling the
+ functions that take a channel argument.
+
+ * include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c,
+ doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by:
+ nic_bellamy Patches:
+ 2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by
+ nic_bellamy (license 213) Add support for configurable file
+ locking methods. The default is "lockfile", which is the old
+ behavior. There is an additional option, "flock", which is
+ intended for use in situations where the lockfile method will not
+ work, such as with SMB/CIFS mounts.
+
+ * /, configs/indications.conf.sample: Merged revisions 81226 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) |
+ 2 lines Add Russian tones. (closes issue #7953, hanabana)
+ ........
+
+2007-08-28 14:37 +0000 [r81210] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: (closes issue #10579) Reported by: ornati
+ Make sure the called channel during the attended transfer process
+ becomes associated with the calling channel so that the
+ ast_waitfor_* call works properly under epoll.
+
+2007-08-28 14:12 +0000 [r81121-81190] Mark Michelson <mmichelson@digium.com>
+
+ * /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28
+ Aug 2007) | 5 lines Fixes a forwarding problem when using
+ res_config_mysql (closes issue #10573, reported by chrisvaughan,
+ patch suggested by chrisvaughan as well) ........
+
+ * /, apps/app_queue.c: Merged revisions 81158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug
+ 2007) | 5 lines Resolve a potential deadlock. In this case, a
+ single queue is locked, then the queue list. In changethread(),
+ the queue list is locked, and then each individual queue is
+ locked. Under the right circumstances, this could deadlock. As
+ such, I have unlocked the individual queue before locking the
+ queue list, and then locked the queue back after the queue list
+ is unlocked. ........
+
+ * /, channels/chan_agent.c: Merged revisions 81120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27
+ Aug 2007) | 7 lines DTMF begin frames should be ignored so that
+ when an agent acks a call with the '#' key, he doesn't cause a
+ queue's announce file to be interrupted. Also went ahead and did
+ the same for the '*' key and for ending a call. (closes issue
+ #10528, reported by deskhack, patched by me) ........
+
+2007-08-27 20:55 +0000 [r81118] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directed_pickup.c: Enhance Pickup to do native
+ pickupgroup pickup when no arguments are specified (closes issue
+ #10404)
+
+2007-08-27 17:44 +0000 [r81043-81098] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh
+ well ... it's harmless. Merged revisions 81065 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) |
+ 1 line explicity define a variable as a boolean ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) |
+ 3 lines Add a \todo to note that this module leaks most of the
+ memory it allocates on unload and should be fixed (when I'm not
+ in the middle of something else ...). ........
+
+ * /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27
+ Aug 2007) | 11 lines (closes issue #10419) Reported by:
+ mustardman Patches: asterisk-mohposition.diff.txt uploaded by
+ jamesgolovich (license 176) This patch fixes a few problems with
+ music on hold. * Fix issues with starting at the beginning of a
+ file when it shouldn't. * Fix the inuse counter to be decremented
+ even if the class had not been set to be deleted when not in use
+ anymore * Don't arbitrarily limit the number of MOH files to 255
+ ........
+
+2007-08-27 15:03 +0000 [r81013] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6
+ lines (closes issue #10561) Reported by: jesselang Patches:
+ chan_sip-ChannelReload-20080825.patch uploaded by jesselang
+ (license 202) Remove an extra \r\n to make the ChannelReload
+ event conform with every other event. ........
+
+2007-08-27 14:56 +0000 [r81011] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 81010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug
+ 2007) | 3 lines Found a case where the queue's membercount is
+ off. It does not take into account dynamic members on a reload.
+ ........
+
+2007-08-27 13:35 +0000 [r80962-80991] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Remove places that say if no language is
+ specified it will default to english... since on some setups this
+ is untrue.
+
+ * /, main/rtp.c: Merged revisions 80974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4
+ lines (closes issue #10562) Reported by: idkpmiller Correct
+ jitter value output in the CLI to be as expected. ........
+
+ * configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG
+ Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20)
+ Fix up sip.conf sample configuration.
+
+2007-08-26 18:12 +0000 [r80933] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) |
+ 3 lines Remove an extra signal_condition() for the scheduler
+ thread. (closes issue #10564, patch from casper) ........
+
+2007-08-25 17:55 +0000 [r80821-80898] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) |
+ 7 lines Fix some issues with the handling of the scheduler in
+ chan_iax2. Most of the places that scheduled items to be executed
+ by the scheduler thread did not signal the scheduler thread to
+ wake up so that it could recalculate the time until the next
+ action. These changes will make the scheduler thread more
+ responsive and ensure that actions get executed as close to when
+ intended as possible instead of it being possible for very long
+ delays. ........
+
+ * pbx/pbx_dundi.c: localize a variable and remove a duplicate error
+ message
+
+ * apps/app_queue.c: use ast_strlen_zero
+
+ * /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) |
+ 5 lines If dnsmgr is in use, and no DNS servers are available
+ when Asterisk first starts, then don't give up on poking peers.
+ Allow the poke to get rescheduled so that it will work once the
+ dnsmgr is able to resolve the host. (closes issue #10521, patch
+ by jamesgolovich) ........
+
+ * /, main/dsp.c: Merged revisions 80820 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) |
+ 7 lines Improve the debouncing logic in the DTMF detector to fix
+ some reliability issues. Previously, this code used a shift
+ register of hits and non-hits. However, if the start of the digit
+ isn't clean, it is possible for the leading edge detector to miss
+ the digit. These changes replace the flawed shift register logic
+ and also does the debouncing on the trailing edge as well.
+ (closes issue #10535, many thanks to softins for the patch)
+ ........
+
+2007-08-24 20:21 +0000 [r80819] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: Merged revisions 80818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007)
+ | 3 lines A minor correction to the available logic of autofill.
+ If a queue member is paused, they're not really "available" so
+ don't count them as such. Somewhat related to issue #10155
+ ........
+
+2007-08-24 19:50 +0000 [r80817] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Fix documentation for Set (closes issue #10549)
+
+2007-08-24 19:03 +0000 [r80790] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 80789 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1
+ line From a complaint by jmls, I realize that the message in
+ cdr_disposition is unnecessary. To get failure disposition, just
+ return -1; no use having more than one case do that. ........
+
+2007-08-24 18:05 +0000 [r80778] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Add VMWI chan_zap support #9909
+
+2007-08-24 15:53 +0000 [r80751] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug
+ 2007) | 3 lines Fix a possible crash in IMAP voicemail. ........
+
+2007-08-24 15:49 +0000 [r80749] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revisions 80747 via svnmerge ........ r80747 |
+ tilghman | 2007-08-24 10:41:43 -0500 (Fri, 24 Aug 2007) | 2 lines
+ Make the deprecation warning inline with the code, instead of
+ only in documentation (closes issue #10549) ........
+
+2007-08-24 15:42 +0000 [r80748] Steve Murphy <murf@digium.com>
+
+ * utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well
+
+2007-08-24 15:29 +0000 [r80725] Russell Bryant <russell@digium.com>
+
+ * /, utils/ael_main.c: Merged revisions 80722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) |
+ 3 lines Tweak the formatting of this MODULEINFO block. I think
+ this would have caused a "*" to get in the menuselect-tree file.
+ ........
+
+2007-08-24 14:55 +0000 [r80690-80718] Steve Murphy <murf@digium.com>
+
+ * /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via
+ svnmerge from