https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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.
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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)
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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.
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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
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r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | 5 lines
When executing a dynamic feature, don't look it up a second time by digit pattern
after we already looked it up by name. This causes broken behavior if there is
more than one feature defined with the same digit pattern.
(closes issue #10539, reported by bungalow, patch by me)
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r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | 5 lines
Juggie in #asterisk-dev was reporting problems where fgets would return
without reading the whole line when using fastagi. When this happens,
errno was set to EINTR or EAGAIN. This patch accounts for the possibility
and lets fgets continue in that case.
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r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16 Aug 2007) | 14 lines
This patch fixes a bug where reloading the module with "module reload" did not
delete classes from memory that were no longer in the config. This patch fixes
that problem as well as another one. Previously, if you reloaded MOH using the
"moh reload" CLI command, which behaved differently than "module reload ...",
MOH had to be stopped on every channel and started again immediately. However,
there was no way to tell what class was being used, so they would all fall back
to the default class.
(closes issue #10139)
Reported by: blitzrage
Patches:
asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich
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r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007) | 21 lines
A fix for two critical problems detected while working with Daniel
McKeehan in issue #10184.
Upon priority change, the resource list is not NULL terminated when
moving an item to the end of the list. This makes Asterisk endlessy
loop whenever it needs to read the list. Jids with different resource and
priority values, like in Gmail's and GoogleTalk's jabber clients put
that problem in evidence.
Upon reception of a 'from' attribute with an empty resource string,
Asterisk crashes when trying to access the found->cap pointer if the
resource list for the given buddy is not empty. This situation is
perfectly valid and must be handled. The Gizmoproject's jabber client
put that problem in evidence.
Also added a few comments in the code as well as a handle for the
capabilities from Gmail's jabber client, which are stored in a caps:c tag
rather than the usual c tag.
Closes issue #10184.
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The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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* Use a thread local ast_str for building the string that will be written out
to the console for debug, and to the FD for the AGI itself, instead of allocating
a buffer on the heap every time the function is called.
* Use the information contained within the ast_str to determine how many bytes
need to be written instead of calling strlen().
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r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines
(closes issue #10279)
Reported by: seanbright
Patches:
res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)
Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.
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r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines
(closes issue #10327)
Reported by: kkiely
Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place.
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- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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This merges the trunk only part of the patches from this issue. In 1.4, res_agi
will issue a warning if you try to use DeadAGI on a channel that is not hung up.
Now, in trunk, it just plain won't let you do it.
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r75067 | russell | 2007-07-13 15:10:40 -0500 (Fri, 13 Jul 2007) | 14 lines
Merged revisions 75059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 Jul 2007) | 6 lines
Ensure that adding a user to the list of users of a specific music on hold
class is not done at the same time as any of the other operations on this list
to prevent list corruption. Using the global moh_data lock for this is not
ideal, but it is what is used to protect these lists everywhere else in the
module, and I am only changing what is necessary to fix the bug.
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Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
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r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09 Jul 2007) | 9 lines
(closes issue #10123)
Reported by: blitzrage
Patches submitted by: juggie, qwell, me
Tested by: blitzrage
When trying to find a music on hold class to use, try all of the options,
instead of only the first one that is set. Also, change the MusicOnHold
applications to not hang up on the channel when a class can not be found.
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function to explicitly check for the int return value. Also, make a few
other minor changes such as removing a variable.
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r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) | 5 lines
Pass HOLD and UNHOLD frames to the other channel when they are returned from a
native bridge function. This fixes a problem where when two zap channels are
natively bridged and one does a flash hook, the other channel did not receive
music on hold. (Reported to me directly by Doug Bailey at Digium)
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for extracting application, function, manager, and agi documentation is the wrong
one to take. The most severe problem is that the output depends on which modules
are loaded as well as compile time options, which both determine which parts are
available.
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r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | 6 lines
To prevent 92138749238754 more reports of "I have unixodbc installed, but
still can't build *_odbc.so!", check for ltdl directly, instead of just listing
it as another library to include in the unixodbc check in the configure script.
This also makes ltdl show up as a dependency in menuselect so people know what
to go install. (related to issue #9989, patch by me)
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the beginning of the file. Also, add a channel variable that indicates
the location in the file where the Playback was stopped.
(closes issue #7655, patch from sharkey)
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r68028 | oej | 2007-06-07 11:55:13 +0200 (Thu, 07 Jun 2007) | 4 lines
Ok, we found out that this is not about if you have any *active* clients using TLS, but
if you have initialized TLS at all during the lifetime of the module. So if you reload
to disable TLS, it won't help.
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r68027 | oej | 2007-06-07 11:42:26 +0200 (Thu, 07 Jun 2007) | 8 lines
If you have a jabber client that uses TLS, refuse unload. Bad fix, but will prevent
crashes while we are trying to find a workaround.
Iksemel development seems to have stalled and we might have to stop using the
TCP/TLS connections in that library and use our own, which would scale better
from a poll/select perspective I guess. It would also make it easier to migrate
to OpenSSL and stop Asterisk from depending on both OpenSSL and GnuTLS.
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r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 lines
Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks!
Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
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r67872 | russell | 2007-06-06 17:08:02 -0500 (Wed, 06 Jun 2007) | 6 lines
Disable reload functionality in res_snmp. It is not possible to initialize the
snmp library more than once without completely unloading the module and loading
it again.
(issue #9571, reported by hristo, additional helpful debug information from festr,
patch from me)
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r67064 | file | 2007-06-04 13:41:59 -0400 (Mon, 04 Jun 2007) | 2 lines
Returning a value that indicates the parking of a call was a success when it really wasn't (because the parking slot selected was in use) is the wrong thing to do. (issue #9723 reported by mdu113)
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features.conf. This allows you to create a feature one time, and then map it
into groups for various different key mappings for the same feature, as well
as easy access control to groups of features.
(patch from bbryant)
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class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
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r62797 | kpfleming | 2007-05-02 19:57:23 -0400 (Wed, 02 May 2007) | 7 lines
improve static Realtime config loading from PostgreSQL:
don't request sorting on fields that are pointless to sort on
use ast_build_string() instead of snprintf()
don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway
(patch by me, inspired by blitzrage's bug report about res_config_odbc)
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r62807 | kpfleming | 2007-05-02 20:02:57 -0400 (Wed, 02 May 2007) | 15 lines
Merged revisions 62796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 May 2007) | 7 lines
increase reliability and efficiency of static Realtime config loading via ODBC:
don't request fields we aren't going to use
don't request sorting on fields that are pointless to sort on
explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
(reported by blitzrage in person (!), patch by me)
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add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)
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r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines
Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines
Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338, patch by svanlund)
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
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r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) | 6 lines
When building a response to a subscription, the "from" must be the full Jabber
ID. This fixes some problems where jabber users are not able to add their
Asterisk account to their user list, since they are unable to get Asterisk
to approve their subscription. (issue #8210, reported by caspy, and verified
by bradtem)
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
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previously set are erroneously still set (Bug 6701). After discussion,
it was determined this should only be changed in trunk.
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r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 lines
Save 1 whopping byte of allocated memory!
This looks like it may have been a chicken/egg scenario..
You had to call a cleanup func, because everything was allocated.
Then since you had to call a cleanup func, you were forced to allocate - ie; strdup("").
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48375 | tilghman | 2006-12-10 18:47:21 -0600 (Sun, 10 Dec 2006) | 13 lines
Merged revisions 48374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines
When doing a fork() and exec(), two problems existed (Issue 8086):
1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(), causing Asterisk
signal handlers within the child to execute, which caused nasty race conditions.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47239 | russell | 2006-11-06 20:25:10 -0500 (Mon, 06 Nov 2006) | 13 lines
Merged revisions 47238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines
If random order is enabled for files mode music on hold, set a random initial
position, instead of always starting at the first file, and doing the random
operation only when switching to the next file.
(bug reported by John Lange on the asterisk-dev mailing list)
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r46778 | russell | 2006-11-01 13:26:35 -0500 (Wed, 01 Nov 2006) | 17 lines
Merged revisions 46776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines
soxmix and Asterisk expect different file extensions for certain formats. This
was already handled for the wav49 format. However, it was not handled for
ulaw and alaw. I fixed this in such a way that using the alternate extensions
for ulaw and alaw will only happen if we know we're calling soxmix, and not a
custom script defined using the MONITOR_EXEC variable. The wav49 processing
was left alone so that external scripts will see no behavior change.
(issue #7550, reported by mnicholson, proposed patch by junky, committed fix
is a bit different)
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r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines
We should always be using _exit() after a fork() or vfork() instead of exit().
This is because exit() does some extra cleanup which in some implementations
of vfork(), for example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
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