- 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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91409 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91408 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91347 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91197 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91077 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91011 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90949 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90433 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90351 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90146 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90060 f38db490-d61c-443f-a65b-d21fe96a405b
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89915 f38db490-d61c-443f-a65b-d21fe96a405b
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().
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89791 f38db490-d61c-443f-a65b-d21fe96a405b
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89721 f38db490-d61c-443f-a65b-d21fe96a405b
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89698 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89612 f38db490-d61c-443f-a65b-d21fe96a405b
- 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?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89578 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89541 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89537 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89490 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89487 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89483 f38db490-d61c-443f-a65b-d21fe96a405b
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89458 f38db490-d61c-443f-a65b-d21fe96a405b
This is only to complete the build, clearly the linker
behaviour will be completely different and likely to
cause trouble in those cases.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89367 f38db490-d61c-443f-a65b-d21fe96a405b
oej