Jeff Peeler [Wed, 11 Jun 2008 16:11:40 +0000 (16:11 +0000)]
Merged revisions 121804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line
add instructions for logging gdb output via set logging on
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Christian Richter [Wed, 11 Jun 2008 11:52:18 +0000 (11:52 +0000)]
Merged revisions 121751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line
fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken.
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Russell Bryant [Tue, 10 Jun 2008 21:51:06 +0000 (21:51 +0000)]
don't refer to asterisk-events, as that implies that the code was checked out from a branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121716
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Tilghman Lesher [Tue, 10 Jun 2008 21:14:58 +0000 (21:14 +0000)]
Move the table cache routines to res_odbc, so they can be used from other
places (app_voicemail, for example).
(Related to bug #11678)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121683
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Mark Michelson [Tue, 10 Jun 2008 19:52:21 +0000 (19:52 +0000)]
Add an additional sanity check in case an event is passed between
Asterisk boxes with mismatched ie_maps.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121649
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Donny Kavanagh [Tue, 10 Jun 2008 19:03:11 +0000 (19:03 +0000)]
Revision 117802 changed frame.data to frame.data.ptr however codec_ilbc.c was not updated. This resolves that oversight.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121599
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Sean Bright [Tue, 10 Jun 2008 18:35:37 +0000 (18:35 +0000)]
Merged revisions 121596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun 2008) | 6 lines
Fixes a problem with some buggy versions of GNU awk (3.1.3) not liking carriage returns in scripts.
(closes issue #12749)
Reported by: alinux
Tested by: Laureano (on #asterisk-dev), juggie
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Russell Bryant [Tue, 10 Jun 2008 15:12:17 +0000 (15:12 +0000)]
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559
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Russell Bryant [Tue, 10 Jun 2008 14:53:40 +0000 (14:53 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121555
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Joshua Colp [Tue, 10 Jun 2008 14:11:16 +0000 (14:11 +0000)]
Fix issue where session timer headers were present when they should not have been.
(closes issue #12706)
Reported by: falves11
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
Tested by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121503
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Russell Bryant [Tue, 10 Jun 2008 14:06:29 +0000 (14:06 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121501
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Joshua Colp [Tue, 10 Jun 2008 13:36:13 +0000 (13:36 +0000)]
Merged revisions 121495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines
If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed.
(closes issue #12828)
Reported by: ramonpeek
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Joshua Colp [Tue, 10 Jun 2008 12:54:39 +0000 (12:54 +0000)]
Merged revisions 121442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines
Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic.
(closes issue #12815)
Reported by: ramonpeek
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Russell Bryant [Tue, 10 Jun 2008 12:50:07 +0000 (12:50 +0000)]
Update dundi.conf to indicate that the asterisk.conf entityid option can be used
to set the entityid used in DUNDi, as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121441
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Russell Bryant [Tue, 10 Jun 2008 12:48:50 +0000 (12:48 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439
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Russell Bryant [Tue, 10 Jun 2008 00:52:46 +0000 (00:52 +0000)]
Bump up the debug level of a couple of messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121407
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Russell Bryant [Tue, 10 Jun 2008 00:43:06 +0000 (00:43 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121403
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Russell Bryant [Tue, 10 Jun 2008 00:31:30 +0000 (00:31 +0000)]
Properly initialize the cli_ping condition and lock
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121402
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Russell Bryant [Tue, 10 Jun 2008 00:29:01 +0000 (00:29 +0000)]
Change system header includes to be like how it is done in other files
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121401
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Tilghman Lesher [Mon, 9 Jun 2008 22:51:59 +0000 (22:51 +0000)]
Expand RQ_INTEGER type out to multiple types, one for each precision
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121367
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Terry Wilson [Mon, 9 Jun 2008 22:42:08 +0000 (22:42 +0000)]
Initialize the lock and destroy lock and cond in the destructor (thanks, mmichelson)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121365
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Tilghman Lesher [Mon, 9 Jun 2008 19:33:23 +0000 (19:33 +0000)]
Add storage of the useragent in the realtime database.
(Closes AST-38)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121334
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Russell Bryant [Mon, 9 Jun 2008 16:55:15 +0000 (16:55 +0000)]
arbitrary formatting change to test mantis change
(closes issue #12824)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121286
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Russell Bryant [Mon, 9 Jun 2008 16:53:26 +0000 (16:53 +0000)]
arbitrary formatting change to test a mantis change
(closes issue #12824)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121285
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Russell Bryant [Mon, 9 Jun 2008 16:48:26 +0000 (16:48 +0000)]
Minor formatting change to test a mantis change ...
(closes issue #12824)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121284
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Russell Bryant [Mon, 9 Jun 2008 16:37:08 +0000 (16:37 +0000)]
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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Tilghman Lesher [Mon, 9 Jun 2008 16:35:06 +0000 (16:35 +0000)]
Implement FINDLABEL matching for the new extension matching engine.
(closes issue #12800)
Reported by: chris-mac
Patches:
20080608__bug12800.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121279
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Mark Michelson [Mon, 9 Jun 2008 15:08:58 +0000 (15:08 +0000)]
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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Michiel van Baak [Sun, 8 Jun 2008 11:40:44 +0000 (11:40 +0000)]
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121197
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Jeff Peeler [Sun, 8 Jun 2008 01:41:59 +0000 (01:41 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121163
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Jeff Peeler [Sun, 8 Jun 2008 01:16:25 +0000 (01:16 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121131
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Russell Bryant [Sat, 7 Jun 2008 14:18:44 +0000 (14:18 +0000)]
Merged revisions 121078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines
Don't run LIST_HEAD_DESTROY on a STATIC list
(closes issue #12807)
Reported by: ys
Patches:
chan_agent_local.diff uploaded by ys (license 281)
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Tilghman Lesher [Fri, 6 Jun 2008 20:24:11 +0000 (20:24 +0000)]
Added a facility for sending arbitrary SIP notify commands from AMI.
(closes issue #12562)
Reported by: michael-fig
Patches:
20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121042
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Tilghman Lesher [Fri, 6 Jun 2008 19:55:08 +0000 (19:55 +0000)]
Make extension match characters case-insensitive.
(closes issue #12777)
Reported by: jsmith
Patches:
lower_case_patterns-trunk-v1.patch uploaded by jsmith (license 15)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121010
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Jeff Peeler [Fri, 6 Jun 2008 18:30:17 +0000 (18:30 +0000)]
Merged revisions 120959 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line
add another LOW_MEMORY define I forgot
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Jeff Peeler [Fri, 6 Jun 2008 18:06:06 +0000 (18:06 +0000)]
Merged revisions 120908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line
only define thread storage variable if necessary for LOW_MEMORY
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Jeff Peeler [Fri, 6 Jun 2008 17:50:05 +0000 (17:50 +0000)]
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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Tilghman Lesher [Fri, 6 Jun 2008 17:34:21 +0000 (17:34 +0000)]
For the purpose of making the changed syntax to ExecIf easier to transition,
allow the deprecated syntax (fixed for jmls on -dev).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120904
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Steve Murphy [Thu, 5 Jun 2008 21:34:42 +0000 (21:34 +0000)]
a small fix for a crash that occurs when compiling AEL with global vars
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120828
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Tilghman Lesher [Thu, 5 Jun 2008 19:07:27 +0000 (19:07 +0000)]
Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789
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Russell Bryant [Thu, 5 Jun 2008 18:03:15 +0000 (18:03 +0000)]
revert 120733, wrong branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120734
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Russell Bryant [Thu, 5 Jun 2008 18:02:37 +0000 (18:02 +0000)]
Update file names
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120733
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Russell Bryant [Thu, 5 Jun 2008 18:01:45 +0000 (18:01 +0000)]
Blocked revisions 120731 via svnmerge
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r120731 | russell | 2008-06-05 13:01:25 -0500 (Thu, 05 Jun 2008) | 2 lines
Add the UPGRADE.txt file from Asterisk 1.2, for handy reference.
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Philippe Sultan [Thu, 5 Jun 2008 17:02:39 +0000 (17:02 +0000)]
Merged revisions 120675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines
Ignore appended resource when comparing JIDs.
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Brett Bryant [Thu, 5 Jun 2008 16:41:36 +0000 (16:41 +0000)]
Update CHANGES file for the things done in revision 120635.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120673
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Russell Bryant [Thu, 5 Jun 2008 16:39:25 +0000 (16:39 +0000)]
Blocked revisions 120671 via svnmerge
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r120671 | russell | 2008-06-05 11:38:52 -0500 (Thu, 05 Jun 2008) | 5 lines
It turns out that searching on the forwarding station isn't very useful for
most people, so pull in the changes that allow searching for SMDI messages
based on other components of the SMDI message. Also, update the SMDI
documentation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120672
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Brett Bryant [Thu, 5 Jun 2008 16:24:19 +0000 (16:24 +0000)]
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635
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Tilghman Lesher [Thu, 5 Jun 2008 15:58:11 +0000 (15:58 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120602
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Tilghman Lesher [Thu, 5 Jun 2008 14:35:47 +0000 (14:35 +0000)]
Add info on the [compat] section of asterisk.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120567
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Mark Michelson [Wed, 4 Jun 2008 22:07:37 +0000 (22:07 +0000)]
Merged revisions 120513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun 2008) | 6 lines
Make sure that the string we set will survive the unref of
the queue member.
Thanks to Russell, who pointed this out.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120514
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Tilghman Lesher [Wed, 4 Jun 2008 20:34:52 +0000 (20:34 +0000)]
MSet doesn't necessarily need chan to be set
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120477
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Tilghman Lesher [Wed, 4 Jun 2008 18:37:08 +0000 (18:37 +0000)]
Merged revisions 120425 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) | 6 lines
If we fail to setup the PRI request channel, don't continue, exit with an error.
(closes issue #11989)
Reported by: Corydon76
Patches:
20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14)
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Russell Bryant [Wed, 4 Jun 2008 16:28:37 +0000 (16:28 +0000)]
Blocked revisions 120371 via svnmerge
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r120371 | russell | 2008-06-04 11:26:43 -0500 (Wed, 04 Jun 2008) | 4 lines
Make the "dialplan remove include" CLI command actually work. Also, tweak
some formatting, and make the success message a little bit more clear.
(closes AST-52)
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Joshua Colp [Wed, 4 Jun 2008 15:38:00 +0000 (15:38 +0000)]
We like tabs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120337
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Mark Michelson [Wed, 4 Jun 2008 14:12:45 +0000 (14:12 +0000)]
Merged revisions 120285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun 2008) | 7 lines
Tab completion when removing a member should give the member's interface,
not the name, since the interface is what is expected for the command.
(closes issue #12783)
Reported by: davevg
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Joshua Colp [Wed, 4 Jun 2008 13:33:59 +0000 (13:33 +0000)]
Merged revisions 120282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 lines
Fix a log message and add a message for when the dialplan is done reloading.
(closes issue #12716)
Reported by: chappell
Patches:
dialplan_reload_2.diff uploaded by chappell (license 8)
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Tilghman Lesher [Tue, 3 Jun 2008 23:17:33 +0000 (23:17 +0000)]
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120230
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Tilghman Lesher [Tue, 3 Jun 2008 22:42:03 +0000 (22:42 +0000)]
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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Jeff Peeler [Tue, 3 Jun 2008 22:17:07 +0000 (22:17 +0000)]
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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Tilghman Lesher [Tue, 3 Jun 2008 22:05:16 +0000 (22:05 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171
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Russell Bryant [Tue, 3 Jun 2008 21:35:11 +0000 (21:35 +0000)]
Merged revisions 120168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
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Mark Michelson [Tue, 3 Jun 2008 21:22:52 +0000 (21:22 +0000)]
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120166
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Russell Bryant [Tue, 3 Jun 2008 19:48:37 +0000 (19:48 +0000)]
Use proper return values for a few application modules
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120129
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Russell Bryant [Tue, 3 Jun 2008 18:43:01 +0000 (18:43 +0000)]
fix build for non debug threads
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120066
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Russell Bryant [Tue, 3 Jun 2008 18:26:51 +0000 (18:26 +0000)]
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120064
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Tilghman Lesher [Tue, 3 Jun 2008 18:24:14 +0000 (18:24 +0000)]
Blocked revisions 120061 via svnmerge
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r120061 | tilghman | 2008-06-03 13:23:32 -0500 (Tue, 03 Jun 2008) | 8 lines
When listing the manager users, managers in users.conf are not shown, even
though they are allowed to connect.
(closes issue #12594)
Reported by: bkruse
Patches:
12594-managerusers-2.diff uploaded by qwell (license 4)
Tested by: bkruse
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Tilghman Lesher [Tue, 3 Jun 2008 16:19:35 +0000 (16:19 +0000)]
Merged revisions 120001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines
Save the callno when we're poking, because our peer structure could change
during destruction (and thus we unlock the wrong callno, causing a
cascade failure).
(closes issue #12717)
Reported by: gewfie
Patches:
20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
Tested by: gewfie
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Steve Murphy [Tue, 3 Jun 2008 15:49:34 +0000 (15:49 +0000)]
Merged revisions 119966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines
Updated the regressions on AEL. Hadn't updated
this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because
it adds extra priorities, and at least one needed to be updated
because of the removal of the empty extension warning
message.
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Steve Murphy [Tue, 3 Jun 2008 15:07:20 +0000 (15:07 +0000)]
Merged revisions 119929 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines
as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
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Joshua Colp [Tue, 3 Jun 2008 14:47:54 +0000 (14:47 +0000)]
Merged revisions 119926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines
Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox)
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Russell Bryant [Tue, 3 Jun 2008 13:29:16 +0000 (13:29 +0000)]
Do a deep copy of file and function strings to avoid a potential crash when
modules are unloaded.
(closes issue #12780)
Reported by: ys
Patches:
logger.diff uploaded by ys (license 281)
-- modified by me for coding guidelines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119892
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Russell Bryant [Mon, 2 Jun 2008 20:08:24 +0000 (20:08 +0000)]
Merged revisions 119838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
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Russell Bryant [Mon, 2 Jun 2008 16:14:15 +0000 (16:14 +0000)]
Add app_fax from asterisk-addons, with some additional changes to resolve compiler
warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being
distributed under the LGPL, so we can move this module into the main tree.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119801
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Russell Bryant [Mon, 2 Jun 2008 15:57:43 +0000 (15:57 +0000)]
After determining that the version of spandsp installed is an acceptable version,
do a build and link test to ensure that the library is usable, and that libtiff
is also available
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119799
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Russell Bryant [Mon, 2 Jun 2008 15:43:40 +0000 (15:43 +0000)]
Add a configure script check for spandsp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119795
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Russell Bryant [Mon, 2 Jun 2008 14:41:55 +0000 (14:41 +0000)]
Merged revisions 119742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines
Improve CLI command blacklist checking for the command manager action. Previously,
it did not handle case or whitespace properly. This made it possible for blacklisted
commands to get executed anyway.
(closes issue #12765)
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Philippe Sultan [Mon, 2 Jun 2008 14:35:24 +0000 (14:35 +0000)]
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119741
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Russell Bryant [Mon, 2 Jun 2008 12:30:42 +0000 (12:30 +0000)]
Merged revisions 119687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines
Even of the first PING or LAGRQ doesn't get sent because it comes up too soon,
make sure to reschedule so it gets sent later.
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Christian Richter [Mon, 2 Jun 2008 09:35:04 +0000 (09:35 +0000)]
Merged revisions 119636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line
fixed compile issue when dev-mode is enabled
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Christian Richter [Mon, 2 Jun 2008 08:46:23 +0000 (08:46 +0000)]
Merged revisions 119585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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Russell Bryant [Mon, 2 Jun 2008 01:08:16 +0000 (01:08 +0000)]
Merged revisions 119533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines
Change a debug message to an actual debug message
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Russell Bryant [Mon, 2 Jun 2008 01:04:01 +0000 (01:04 +0000)]
Merged revisions 119530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines
Fix another typo in documentation
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Michiel van Baak [Sun, 1 Jun 2008 21:06:27 +0000 (21:06 +0000)]
Merged revisions 119478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines
small typo fix 'retires' => 'retries'
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Russell Bryant [Fri, 30 May 2008 21:51:17 +0000 (21:51 +0000)]
Fix a minor merge issue that caused a function to not get compiled in with
DEBUG_THREADS like it was supposed to
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119423
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Tilghman Lesher [Fri, 30 May 2008 21:23:14 +0000 (21:23 +0000)]
Merged revisions 119404 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines
When joinempty=strict, it only failed on join if there were busy members. If
all members were logged out OR paused, then it (incorrectly) let callers join
the queue.
(closes issue #12451)
Reported by: davidw
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Joshua Colp [Fri, 30 May 2008 19:47:30 +0000 (19:47 +0000)]
Merged revisions 119354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 lines
Fix a bug I found while testing for another issue.
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Michiel van Baak [Fri, 30 May 2008 16:47:24 +0000 (16:47 +0000)]
Merged revisions 119301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) | 14 lines
dont use a bashism way to check the $VERSION variable.
The rc/init.d scripts, and safe_asterisk work on normal sh now again.
Tested on:
OpenBSD 4.2 (me)
Debian etch (me)
Ubuntu Hardy (me and loloski)
FC9 (loloski)
(closes issue #12687)
Reported by: loloski
Patches:
20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7)
Tested by: loloski, mvanbaak
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Tilghman Lesher [Fri, 30 May 2008 16:40:13 +0000 (16:40 +0000)]
Suppress warning about pbx structure already existing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119299
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Tilghman Lesher [Fri, 30 May 2008 16:10:46 +0000 (16:10 +0000)]
Add native AGI command GOSUB, as invoking Gosub with EXEC does not work
properly.
(closes issue #12760)
Reported by: Corydon76
Patches:
20080530__bug12760.diff.txt uploaded by Corydon76 (license 14)
Tested by: tim_ringenbach, Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119296
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Russell Bryant [Fri, 30 May 2008 12:59:11 +0000 (12:59 +0000)]
Merged revisions 119238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines
Merged revisions 119237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines
- Instead of only enforcing destination call number checking on an ACK, check
all full frames except for PING and LAGRQ, which may be sent by older versions
too quickly to contain the destination call number.
(As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
from being sent before the destination call number is known.
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Olle Johansson [Fri, 30 May 2008 11:26:05 +0000 (11:26 +0000)]
Prefer T140 with REDundance before T140 without.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119207
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Russell Bryant [Thu, 29 May 2008 22:28:50 +0000 (22:28 +0000)]
Merged revisions 119156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines
Fix a race condition in channel autoservice. There was still a small window of opportunity
for a DTMF frame, or some other deferred frame type, to come in and get dropped.
(closes issue #12656)
(closes issue #12656)
Reported by: dimas
Patches:
v3-12656.patch uploaded by dimas (license 88)
-- with some modifications by me
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119157
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Brett Bryant [Thu, 29 May 2008 21:30:37 +0000 (21:30 +0000)]
Adds support for changing logger settingss on remote consoles with a
new command "logger set level".
i.e. "logger set level debug off"
(closes issue #10891)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119126
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Russell Bryant [Thu, 29 May 2008 20:49:48 +0000 (20:49 +0000)]
Blocked revisions 119076 via svnmerge
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r119076 | russell | 2008-05-29 15:48:33 -0500 (Thu, 29 May 2008) | 3 lines
Oddly enough, all of the contents of audiohook.h were in there twice. I have
removed the second copy.
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Steve Murphy [Thu, 29 May 2008 20:26:42 +0000 (20:26 +0000)]
Had to move the ASTERISK_FILE_VERSION decl to just after the include of "asterisk.h"
or you get undefined variable errors when you are compiling under the influence
of MTX_PROFILE
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119074
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Tilghman Lesher [Thu, 29 May 2008 20:25:33 +0000 (20:25 +0000)]
Merged revisions 119071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines
Call waiting tone occurs too often, because it's getting serviced by both
subchannels.
(closes issue #11354)
Reported by: cahen
Patches:
20080512__bug11354.diff.txt uploaded by Corydon76 (license 14)
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Michiel van Baak [Thu, 29 May 2008 19:10:31 +0000 (19:10 +0000)]
Make sure the nrfds and nefds are reset to NULL
before we enter manage_parkinglot.
This will get rid of CLI warnings like:
__ast_read: Exception flag set on 'SIP/<NUMBER>-<ID>', but no exception handler
(closes issue #12748)
Reported by: nreinartz
Patches:
asterisk-multiparking_initialize_filedescr_sets-0.0.1.patch uploaded by nreinartz (license 452)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119015
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Russell Bryant [Thu, 29 May 2008 19:05:33 +0000 (19:05 +0000)]
Merged revisions 119012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) | 4 lines
- Fix a typo in the argument to Playtones
- use ast_safe_sleep() instead of calling the wait application
(thanks to tilghman for pointing these out!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119013
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Russell Bryant [Thu, 29 May 2008 18:54:11 +0000 (18:54 +0000)]
Merged revisions 119009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines
Merged revisions 119008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines
Merge changes from team/russell/iax2-another-fix-to-the-fix
As described in the following post to the asterisk-dev mailing list, only
enforce destination call numbers when processing an ACK.
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
(closes issue #12631)
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Russell Bryant [Thu, 29 May 2008 17:52:00 +0000 (17:52 +0000)]
Merged revisions 118961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) | 3 lines
- Mark app_milliwatt dependent on res_indications (thanks to jsmith)
- fix a typo in a log message (thanks to qwell)
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