git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

net result of this work is that attended transfers made
by queue members will now show up in the queue_log as a 
TRANSFER message instead of COMPLETECALLER as it had been.

As far as the details go, I created a datastore which is
attached to the calling channel just prior to when the caller
is bridged with the queue member. If the calling channel
is masqueraded, then during the "fixup" portion, the TRANSFER
will be logged and the datastore will be removed.



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines

Flipflop the sections for two options, since the section for 'X' (exit context)
may otherwise absorb keypresses meant for 's' (admin/user menu).
(closes issue #12836)
 Reported by: blitzrage
 Patches: 
       20080611__bug12836.diff.txt uploaded by Corydon76 (license 14)
 Tested by: blitzrage

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r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines

Occasionally, the alertpipe loses its nonblocking status, so detect and correct
that situation before it causes a deadlock.  (Reported and tested by ctooley
via #asterisk-dev)

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r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line

Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines

(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia

Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.

The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.

The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.

The T option was added to forkCDR to force 
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.

The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via 
email, irc, etc, over the past months/year)

The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.

Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

add infrastructure so that timing source can be a loadable module... next steps are to convert channel.c and chan_iax2.c to use this new API, and to move all the DAHDI-specific timing source code into a new res_timing_dahdi module


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those that do can check their setup using CLI commands.
(closes issue #12839)


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r121992 | twilson | 2008-06-11 18:47:23 -0500 (Wed, 11 Jun 2008) | 2 lines

Backport fix for 11520--for some reason I didn't do this back in February when I patched for trunk.

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines

Make calls to ast_assert() actually test something, so that the error message
printed is not nonsensical (reported by mvanbaak via #asterisk-bugs).

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Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message.  Fixes crash bug

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes issue #12831)


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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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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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places (app_voicemail, for example).
(Related to bug #11678)


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Asterisk boxes with mismatched ie_maps.



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Revision 117802 changed frame.data to frame.data.ptr however codec_ilbc.c was not updated.  This resolves that oversight.


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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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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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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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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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to set the entityid used in DUNDi, as well.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121441 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes AST-38)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121334 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12824)


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(closes issue #12824)


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(closes issue #12824)


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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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