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(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/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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to realtime less painful in the future.
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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since another thread could remove them.
(closes issue #12541)
Reported by: snuffy
Patches:
bug_12156_apps.diff uploaded by snuffy (license 35)
Several additional changes by me
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yet or not
(Closes issue #12292)
Reported by: DEA
Patches:
app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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a Meetme conference.
(closes issue #11691)
Reported by: junky
Patches:
meetme_info.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes issue #12430)
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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/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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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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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/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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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11236)
Reported by: philipps
Patches:
20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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