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r121649 | mmichelson | 2008-06-10 14:52:21 -0500 (Tue, 10 Jun 2008) | 4 lines
Add an additional sanity check in case an event is passed between
Asterisk boxes with mismatched ie_maps.
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r121559 | russell | 2008-06-10 10:12:17 -0500 (Tue, 10 Jun 2008) | 24 lines
Merge another big set of changes from team/russell/events
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
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r121555 | russell | 2008-06-10 09:53:40 -0500 (Tue, 10 Jun 2008) | 22 lines
Merge some more changes from team/russell/events
This commit pulls in a batch of improvements and additions to the event API.
Changes include:
- the ability to dynamically build a subscription. This is useful if you're
building a subscription based on something you receive from the network,
or from options in a configuration file.
- Add tables of event types and IE types and the corresponding string
representation for implementing text based protocols that use these
events, for showing events on the CLI, reading configuration that
references event information, among other things.
- Add a table that maps IE types and the corresponding payload type.
- an API call to get the total size of an event
- an API call to get all events from the cache that match a subscription
- a new IE payload type, raw, which I used for transporting the Entity ID in
my code for handling distributed device state.
- Code improvements to reduce code duplication
- Include the Entity ID of the server that originated the event in every event
- an additional event type, DEVICE_STATE_CHANGE, to help facilitate distributed
device state. DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE
is the aggregate device state change across all servers.
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r121501 | russell | 2008-06-10 09:06:29 -0500 (Tue, 10 Jun 2008) | 9 lines
Merge another change from team/russell/events
This commit breaks out some logic from pbx.c into a simple API. The hint
processing code had logic for taking the state from multiple devices and
turning that into the state for a single extension. So, I broke this out
and made an API that lets you take multiple device states and determine
the aggregate device state. I needed this for some core device state changes
to support distributed device state.
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r121439 | russell | 2008-06-10 07:48:50 -0500 (Tue, 10 Jun 2008) | 13 lines
Merge another change from team/russell/events ...
DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
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r121367 | tilghman | 2008-06-09 17:51:59 -0500 (Mon, 09 Jun 2008) | 2 lines
Expand RQ_INTEGER type out to multiple types, one for each precision
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r121403 | russell | 2008-06-09 19:43:06 -0500 (Mon, 09 Jun 2008) | 4 lines
Merge a couple of configure script checks in from team/russell/events. This adds
the checks for the CLM and EVT services from the SAForum AIS. I'm going to work
on merging in changes from this branch in pieces.
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r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines
Merged revisions 121280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines
Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
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r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines
Merged revisions 121229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
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r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
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r121197 | mvanbaak | 2008-06-08 13:40:44 +0200 (Sun, 08 Jun 2008) | 12 lines
add a new argument to PrivacyManager to specify a context
where the entered phone number is checked.
You can now define a set of extensions/exten patterns that describe
valid phone numbers. PrivacyManager will check that context for a match
with the given phone number.
This way you get better control. For example people blindly hitting
10 digits just to get past privacymanager
Example line in extensions.conf:
exten => incoming,n,PrivacyManager(3,10,route-outgoing)
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r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines
This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
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r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines
Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables.
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r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines
Merged revisions 120863,120885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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r120789 | tilghman | 2008-06-05 14:07:27 -0500 (Thu, 05 Jun 2008) | 3 lines
Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future.
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r120673 | bbryant | 2008-06-05 11:41:36 -0500 (Thu, 05 Jun 2008) | 1 line
Update CHANGES file for the things done in revision 120635.
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r120635 | bbryant | 2008-06-05 11:24:19 -0500 (Thu, 05 Jun 2008) | 14 lines
This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
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r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines
Conditionally load the AGI command gosub, depending on whether or not res_agi
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
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r120230 | tilghman | 2008-06-03 18:17:33 -0500 (Tue, 03 Jun 2008) | 7 lines
Add a function, CHANNELS(), which retrieves a list of all active channels.
(closes issue #11330)
Reported by: rain
Patches:
func_channel-channel_list_function.diff uploaded by rain (license 327)
(with some additional changes by me, mostly to meet coding guidelines)
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r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines
Merged revisions 120226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines
Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
Reported by: chappell
Patches:
pbx_loopback.c.diff uploaded by chappell (license 8)
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r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines
Merged revisions 120173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines
(closes issue #11594)
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
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r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines
Move compatibility options into asterisk.conf, default them to on for upgrades,
and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
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r120166 | mmichelson | 2008-06-03 16:22:52 -0500 (Tue, 03 Jun 2008) | 13 lines
Adding two new queue log events. The ADDMEMBER event is logged when
a dynamic realtime queue member is added to the queue, and the
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.
(closes issue #12774)
Reported by: atis
Patches:
queue_log_rt_members.patch uploaded by atis (license 242)
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r120129 | russell | 2008-06-03 14:48:37 -0500 (Tue, 03 Jun 2008) | 2 lines
Use proper return values for a few application modules
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r120064 | russell | 2008-06-03 13:26:51 -0500 (Tue, 03 Jun 2008) | 10 lines
Add lock tracking for rwlocks. Previously, lock.h only had the ability to
hold tracking information for mutexes. Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.
(closes issue #11279)
Reported by: ys
Patches:
trunk_lock_utils.v8.diff uploaded by ys (license 281)
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