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https://origsvn.digium.com/svn/asterisk/trunk
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r134260 | kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 lines
build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI
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https://origsvn.digium.com/svn/asterisk/trunk
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r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines
merging the zap_and_dahdi_trunk branch up to trunk
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https://origsvn.digium.com/svn/asterisk/trunk
Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which
also solved problems with contexts having
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.
Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
(at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
was forked, making special note of all revs that
were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
determine if it was involved in the merge_and_delete
enhancements in trunk. 25 commits were done in 1.6.0,
all but one (106306) was a merge from trunk.
Trunk had 22 additional changes, of which 7 were
involved in the merge_and_delete enhancements:
106757
108894
109169
116461
123358
130145
130297
f. Go to trunk and collect patches, one by one,
of the changes made by each rev across the
entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and
resolve all failures and compilation problems
before proceding to the next patch.
h. test the stuff.
i. profit!
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r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines
(closes issue #13041)
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
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https://origsvn.digium.com/svn/asterisk/trunk
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r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines
Janitor project to convert sizeof to ARRAY_LEN macro.
(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
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https://origsvn.digium.com/svn/asterisk/trunk
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r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines
Merged revisions 125132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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https://origsvn.digium.com/svn/asterisk/trunk
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r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) | 16 lines
Merged revisions 122137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines
Flipflop the sections for two options, since the section for 'X' (exit context)
may otherwise absorb keypresses meant for 's' (admin/user menu).
(closes issue #12836)
Reported by: blitzrage
Patches:
20080611__bug12836.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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https://origsvn.digium.com/svn/asterisk/trunk
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r114617 | tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines
Fix DST calculation, and fix bug in calculation of whether conf has started yet or not
(Closes issue #12292)
Reported by: DEA
Patches:
app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3)
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https://origsvn.digium.com/svn/asterisk/trunk
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r114096 | tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines
Use ast_mkdir instead of mkdir
(Closes issue #12430)
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https://origsvn.digium.com/svn/asterisk/trunk
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r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | 14 lines
Merged revisions 114029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines
Create the directory where name recordings will go if it does not exist.
(closes issue #12311)
Reported by: rkeene
Patches:
12311-mkdir.diff uploaded by qwell (license 4)
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https://origsvn.digium.com/svn/asterisk/trunk
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r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines
Merged revisions 110163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines
Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.
(closes issue #11990, reported by anakaoka, patched by me)
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https://origsvn.digium.com/svn/asterisk/trunk
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r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines
Merged revisions 107637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines
Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue #12136)
Reported by: aragon
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https://origsvn.digium.com/svn/asterisk/trunk
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r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008) | 10 lines
Merged revisions 106895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines
Only start the SLA thread if SLA has actually been configured.
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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
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(closes issue #11763)
Reported by: IgorG
Patches:
docupd.v1.diff uploaded by IgorG (license 20)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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(closes issue #11658, reported and patched by eliel)
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Also, a bunch of coding guidelines fixes.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11048)
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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.
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MeetMeLeave)
(Moremanager)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #11383)
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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.
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- 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?
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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* 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)
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were included almost everywhere.
Remove some of the instances.
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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.).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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- 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
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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)
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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)
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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!)
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Reported by: eliel
Patch by: eliel
Closes issue #11135
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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(closes issue #10911, reported by junky, patched by me)
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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)
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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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)
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have Command privilege. (closes issue #10736)
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(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)
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