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: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
make the format explicit in a debug message;
print the audio instead of aggregated peer capability in a debugging msg.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
