[svn-commits] branch jcollie/bug7021 r30876 - in /team/jcollie/bug7021: ./ apps/ build_tool...

[svn-commits at lists.digium.com]

Author: jcollie
Date: Tue May 30 14:35:02 2006
New Revision: 30876

URL: http://svn.digium.com/view/asterisk?rev=30876&view=rev
Log:
Merged revisions 29553-29554,29556,29570,29592,29619,29641,29665,29667-29668,29703-29708,29727,29731,29734-29735,29765-29766,29803,29825,29846-29848,29850,29880,29903,29905,29935,29938,29970,29972,29988,30010,30012,30034,30036,30038-30040,30068,30070,30099,30104,30131-30132,30152,30173,30194,30216,30240-30242,30271-30272,30294,30297,30299,30328,30337,30359-30361,30384,30390,30409,30411,30426-30427,30430,30458,30463,30465,30490,30521,30547-30548,30578-30580,30603,30607,30630,30653-30655,30677,30699-30701,30723,30746,30771,30800,30803-30806,30835,30837,30847,30875 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r29553 | mogorman | 2006-05-22 16:12:30 -0500 (Mon, 22 May 2006) | 1 line

asterisk-xmpp merge in
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r29554 | kpfleming | 2006-05-22 16:20:01 -0500 (Mon, 22 May 2006) | 2 lines

bootstrap updates to include xmpp related stuff

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r29556 | file | 2006-05-22 16:28:32 -0500 (Mon, 22 May 2006) | 10 lines

Merged revisions 29555 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r29555 | file | 2006-05-22 18:27:12 -0300 (Mon, 22 May 2006) | 2 lines

Increase the silence threshold to 128 to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed by casper)

........

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r29570 | kpfleming | 2006-05-22 17:02:38 -0500 (Mon, 22 May 2006) | 2 lines

remove properties from the xmpp merge

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r29592 | mogorman | 2006-05-22 17:51:56 -0500 (Mon, 22 May 2006) | 3 lines

patch from bug 0007204 to make bug 5750 follow standard
manager naming convention.

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r29619 | markster | 2006-05-22 23:49:31 -0500 (Mon, 22 May 2006) | 2 lines

Handle ringing (early) state properly on SIP

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r29641 | oej | 2006-05-23 06:15:57 -0500 (Tue, 23 May 2006) | 2 lines

Formatting, typos

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r29665 | oej | 2006-05-23 07:14:35 -0500 (Tue, 23 May 2006) | 2 lines

Code formatting

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r29667 | crichter | 2006-05-23 07:38:06 -0500 (Tue, 23 May 2006) | 10 lines

* export_ies uses now _VAR else the vars are not copied to the dest chan
* when receiving a connect from the NT Side we wait until we have the final
  l3id until we queue the answer to asterisk to avoid bridging conflicts
* when not bridged to misdn we had a segfault after receiving the connect 
  due to a strcasecmp bug.. this didn't happen before, cause we hadn't had
  the bridge before
* cleanup of the bchannels is queued now, due to possible race conditions
* added mISDN_clear_stack when cleaning the bchannel


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r29668 | oej | 2006-05-23 07:39:48 -0500 (Tue, 23 May 2006) | 2 lines

Breaking once will stop us... :-)

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r29703 | russell | 2006-05-23 11:25:37 -0500 (Tue, 23 May 2006) | 2 lines

update chan_jingle to reflect the recent change to the indicate prototype

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r29704 | russell | 2006-05-23 11:33:04 -0500 (Tue, 23 May 2006) | 2 lines

remove an unnecessary error message that is really an old debug message

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r29705 | bweschke | 2006-05-23 11:35:46 -0500 (Tue, 23 May 2006) | 3 lines

  app_meetme Muting and Manager API enhancements #6731 (softins w/some minor mods to accomodate recent enum work)


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r29706 | bweschke | 2006-05-23 11:37:40 -0500 (Tue, 23 May 2006) | 11 lines

Merged revisions 29696 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29696 | bweschke | 2006-05-23 11:58:24 -0400 (Tue, 23 May 2006) | 3 lines

 Fix a potential leak and correct (hopefully) a segfault under certain conditions. #6784 (vovan and perry testing)


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r29707 | russell | 2006-05-23 11:37:46 -0500 (Tue, 23 May 2006) | 2 lines

remove another excess "debug" message

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r29708 | mogorman | 2006-05-23 11:43:58 -0500 (Tue, 23 May 2006) | 3 lines

finish cleaning up some more stuff before russell
gets a chance to.

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r29727 | kpfleming | 2006-05-23 12:04:07 -0500 (Tue, 23 May 2006) | 3 lines

restore AST_LIST_HEAD_INIT (with no users in the tree right now)
update ast_mutex_init to allow mutexes that are all zero bytes to be initialized (in the case of a dynamically-allocated structure containing a mutex)

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r29731 | russell | 2006-05-23 12:09:51 -0500 (Tue, 23 May 2006) | 2 lines

on a clean, we have to clean out the ael directory too

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r29734 | bweschke | 2006-05-23 12:21:02 -0500 (Tue, 23 May 2006) | 3 lines

 What's good for 1.2 isn't good for /trunk. Fix for /trunk coming next...


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r29735 | bweschke | 2006-05-23 12:31:05 -0500 (Tue, 23 May 2006) | 3 lines

 Sanity check code for an extended failure in trying to obtain a channel lock that may have been obtained elsewhere. Prevents the monitor thread of the SIP module from going into an infinite loop, effectively, breaking SIP until you restart Asterisk or the mutex is unlocked, whichever comes first.


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r29765 | kpfleming | 2006-05-23 13:17:40 -0500 (Tue, 23 May 2006) | 10 lines

Merged revisions 29764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29764 | kpfleming | 2006-05-23 13:16:40 -0500 (Tue, 23 May 2006) | 2 lines

simplify/fix lock retry, and fix comment

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r29766 | mogorman | 2006-05-23 13:23:05 -0500 (Tue, 23 May 2006) | 3 lines

allows for configurable answer timeout on attended transfer
patch 0006763 with minor changes.

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r29803 | crichter | 2006-05-23 14:40:16 -0500 (Tue, 23 May 2006) | 1 line

added a l1watcher timeout, therefore removed the old behaviour of guessing the l1state.
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r29825 | mogorman | 2006-05-23 15:25:23 -0500 (Tue, 23 May 2006) | 2 lines

fixes bug where server goes away.

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r29846 | mattf | 2006-05-23 16:10:55 -0500 (Tue, 23 May 2006) | 2 lines

Bump up the echo tail length option

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r29847 | mogorman | 2006-05-23 16:18:07 -0500 (Tue, 23 May 2006) | 3 lines

hmm still need a way to get rid of connections
later on.

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r29848 | mogorman | 2006-05-23 16:28:14 -0500 (Tue, 23 May 2006) | 2 lines

get rid of that transport sillyness

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r29850 | russell | 2006-05-23 16:46:26 -0500 (Tue, 23 May 2006) | 10 lines

Merged revisions 29849 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29849 | russell | 2006-05-23 17:44:52 -0400 (Tue, 23 May 2006) | 2 lines

fix the sourceaddress option (issue #7213, alphaque)

........

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r29880 | russell | 2006-05-23 17:57:03 -0500 (Tue, 23 May 2006) | 3 lines

further codec module optimization based on codec_alaw changes by rizzo
(issue #7190, Mithraen)

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r29903 | kpfleming | 2006-05-23 22:28:49 -0500 (Tue, 23 May 2006) | 4 lines

add a new option for 'obscuring' SIP user/peer names from fishers
use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response

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r29905 | kpfleming | 2006-05-23 22:32:55 -0500 (Tue, 23 May 2006) | 2 lines

block SIP obscurity fix from merging... trunk version was different

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r29935 | jdixon | 2006-05-24 02:01:02 -0500 (Wed, 24 May 2006) | 2 lines

Added incoming audio notch filtering, plus a bunch of command improvements, etc.

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r29938 | crichter | 2006-05-24 02:58:52 -0500 (Wed, 24 May 2006) | 1 line

fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
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r29970 | kpfleming | 2006-05-24 11:19:57 -0500 (Wed, 24 May 2006) | 10 lines

Merged revisions 29969 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29969 | kpfleming | 2006-05-24 11:17:26 -0500 (Wed, 24 May 2006) | 2 lines

respect 'usecallingpres' in zapata.conf even if CLID has not been set for the channel (issue #7123)

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r29972 | kpfleming | 2006-05-24 11:54:10 -0500 (Wed, 24 May 2006) | 10 lines

Merged revisions 29971 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29971 | kpfleming | 2006-05-24 11:52:08 -0500 (Wed, 24 May 2006) | 2 lines

fix various bugs related to exiting from queue via keypress and moh handling (issue #6776, different fix)

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r29988 | kpfleming | 2006-05-24 12:02:54 -0500 (Wed, 24 May 2006) | 10 lines

Merged revisions 29973 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r29973 | kpfleming | 2006-05-24 11:59:20 -0500 (Wed, 24 May 2006) | 2 lines

support video recording via AGI 'RECORD FILE' command (issue #7068)

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r30010 | oej | 2006-05-24 13:17:50 -0500 (Wed, 24 May 2006) | 2 lines

Typo fix. Thanks Peter!

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r30012 | kpfleming | 2006-05-24 13:18:14 -0500 (Wed, 24 May 2006) | 2 lines

don't force the compiler name

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r30034 | file | 2006-05-24 14:16:07 -0500 (Wed, 24 May 2006) | 10 lines

Merged revisions 30033 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30033 | file | 2006-05-24 16:14:01 -0300 (Wed, 24 May 2006) | 2 lines

Fix deadlock caused by a race condition in the logger when reloading (issue #7195 reported and fixed by softins)

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r30036 | file | 2006-05-24 14:46:14 -0500 (Wed, 24 May 2006) | 9 lines

Blocked revisions 30035 via svnmerge

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r30035 | file | 2006-05-24 16:44:26 -0300 (Wed, 24 May 2006) | 2 lines

Merge branch for bug 6264 (Privacy option 2 returns dial-status ANSWER / option_priority_jumping not respected) (reported by jkoopmann and branch by murf

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r30038 | oej | 2006-05-24 14:58:36 -0500 (Wed, 24 May 2006) | 2 lines

Typo

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r30039 | mogorman | 2006-05-24 14:59:04 -0500 (Wed, 24 May 2006) | 2 lines

reset timeout on reconnect.

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r30040 | file | 2006-05-24 15:00:10 -0500 (Wed, 24 May 2006) | 2 lines

Merge branch for bug 6264 (Privacy option 2 returns dial-status ANSWER / option_priority_jumping not respected) (reported by jkoopmann and branch by murf)

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r30068 | mogorman | 2006-05-24 15:07:02 -0500 (Wed, 24 May 2006) | 11 lines

Merged revisions 30037 via svnmerge from 
https://svn.digium.com/svn/asterisk/branches/1.2

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r30037 | mogorman | 2006-05-24 14:55:21 -0500 (Wed, 24 May 2006) | 3 lines

app_meemte used the ast_max_exten instead of path_max
solves bug 6822

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r30070 | kpfleming | 2006-05-24 15:37:45 -0500 (Wed, 24 May 2006) | 2 lines

don't send CANCEL for an INVITE that we haven't received a provisional response for yet... mark it pending until a response arrives (issue #7079)

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r30099 | kpfleming | 2006-05-24 16:25:46 -0500 (Wed, 24 May 2006) | 2 lines

block chan_sip fix that is not broken here :-)

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r30104 | file | 2006-05-24 17:00:37 -0500 (Wed, 24 May 2006) | 2 lines

Update some documentation (file internal brain bug #42)

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r30131 | file | 2006-05-24 18:08:39 -0500 (Wed, 24 May 2006) | 2 lines

Nothing to see here... move along

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r30132 | crichter | 2006-05-24 18:21:03 -0500 (Wed, 24 May 2006) | 1 line

added EVENT_NEW_CHANNEL. We change the channel name now when we got the real channel, also changed name generation to new stringfield api
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r30152 | bweschke | 2006-05-24 19:11:30 -0500 (Wed, 24 May 2006) | 3 lines

 Make sure we catch all the instances where a member didn't answer the call sent to them rather than just on a timeout after a dial attempt and some minor code cleanup/reuse.


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r30173 | bweschke | 2006-05-24 20:40:20 -0500 (Wed, 24 May 2006) | 3 lines

 Doxygen comment for Qwell


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r30194 | bweschke | 2006-05-25 08:51:44 -0500 (Thu, 25 May 2006) | 3 lines

 Making sure a char ptr is initialized before we strchr on it is a GOOD thing. Ya! Testing! 


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r30216 | bweschke | 2006-05-25 09:32:15 -0500 (Thu, 25 May 2006) | 3 lines

 Properly initialize destination variables before we send them into pbx_substitute_variables_helper(..). Ya! Testing! Take 2.


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r30240 | file | 2006-05-25 10:29:30 -0500 (Thu, 25 May 2006) | 10 lines

Merged revisions 30239 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30239 | file | 2006-05-25 12:27:44 -0300 (Thu, 25 May 2006) | 2 lines

Get rid of an incorrect SIP dial string in the sample extensions.conf - I even tried variations... no go (issue #7222 reported by arkadia)

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r30241 | russell | 2006-05-25 10:40:38 -0500 (Thu, 25 May 2006) | 3 lines

add DB_DELETE function for the common case of retrieving and deleting a key in
a single operation (issue #7214, twilson)

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r30242 | file | 2006-05-25 10:42:11 -0500 (Thu, 25 May 2006) | 2 lines

Only get the low 16 bits if we actually have a message count

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r30271 | file | 2006-05-25 11:14:17 -0500 (Thu, 25 May 2006) | 2 lines

Safely traverse the thread lists and wait until each thread is done before moving on to the next.

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r30272 | russell | 2006-05-25 11:44:22 -0500 (Thu, 25 May 2006) | 8 lines

- add support for setting an AGISTATUS variable that indicates successful
  execution, failure, or if the channel requested hangup.
- only return -1 from the application if the application requested hangup. If
  there was just a failure in execution of the AGI, just set the status
  variable appropriately and move on in the dialplan.
(issue #7121, original patch by Alessandro Polverini, updated patch by srt,
 committed patch is heavily modified to allow still returning -1 on hangup)

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r30294 | kpfleming | 2006-05-25 12:22:26 -0500 (Thu, 25 May 2006) | 10 lines

Merged revisions 30293 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30293 | kpfleming | 2006-05-25 12:18:01 -0500 (Thu, 25 May 2006) | 2 lines

allow SIPCHANINFO(peername) to work for calls from users as well (issue #7215)

........

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r30297 | kpfleming | 2006-05-25 12:39:46 -0500 (Thu, 25 May 2006) | 10 lines

Merged revisions 30296 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30296 | kpfleming | 2006-05-25 12:39:33 -0500 (Thu, 25 May 2006) | 2 lines

don't try to use -march=s390 when building on S/390 systems (reported via asterisk-users mailing list)

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r30299 | file | 2006-05-25 12:58:55 -0500 (Thu, 25 May 2006) | 2 lines

Add Archive option to call files and add documentation on them. (issue #5426 reported by ezio - props to blitzrage for proof reading the documentation)

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r30328 | russell | 2006-05-25 13:31:19 -0500 (Thu, 25 May 2006) | 5 lines

Add the ability to retrieve the exit code of the forked AGI process.  If there
is an error executing the AGI script, or the AGI script itself returns a
non-zero value, the AGISTATUS variable will now be set to FAILURE instead of
SUCCESS.

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r30337 | russell | 2006-05-25 14:01:26 -0500 (Thu, 25 May 2006) | 4 lines

allow setting a channel variable to disable sending SIGHUP to the AGI process
(issue #6491, original patch by juggie, channel variable patch by corydon,
 committed patch modified to change variable name and update documentation)

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r30359 | russell | 2006-05-25 14:07:31 -0500 (Thu, 25 May 2006) | 6 lines

- mark some applications deprecated that already have replacements
- add BLACKLIST and mark LookupBlacklist deprecated
- add transfercapability support to CHANNEL and mark SetTransferCapability
  deprecated
(issue #7225, Corydon)

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r30360 | russell | 2006-05-25 14:21:09 -0500 (Thu, 25 May 2006) | 3 lines

add the ability to be able to echo DTMF_BEGIN/END, HTML, and IMAGE frames, too 
(issue #7193, Mithraen, with some mods)

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r30361 | kpfleming | 2006-05-25 14:26:26 -0500 (Thu, 25 May 2006) | 2 lines

use the proper method for adding a new entry

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r30384 | file | 2006-05-25 15:05:52 -0500 (Thu, 25 May 2006) | 10 lines

Merged revisions 30373 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30373 | file | 2006-05-25 17:03:11 -0300 (Thu, 25 May 2006) | 2 lines

Don't play the enter sound twice when a person joins a conference after the leader has joined it. (issue #6138 reported by shanermn)

........

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r30390 | file | 2006-05-25 15:51:27 -0500 (Thu, 25 May 2006) | 2 lines

Merge in branch which gives you the ability to set the hangup causecode using the Hangup application. (issue #7160 reported by kmilitzer branch by jcollie)

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r30409 | file | 2006-05-25 16:06:08 -0500 (Thu, 25 May 2006) | 2 lines

Remove possibility of sending duplicate MeetmeJoin manager events, and only send a MeetmeLeave event if a MeetmeJoin event occured in the first place. (issue #6599 reported by imran - provided patch with few tiny mods)

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r30411 | tilghman | 2006-05-25 16:06:43 -0500 (Thu, 25 May 2006) | 2 lines

Deprecate SetCallerID (should have happened prior to release of 1.2)

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r30426 | bweschke | 2006-05-25 16:24:12 -0500 (Thu, 25 May 2006) | 11 lines

Merged revisions 30424 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30424 | bweschke | 2006-05-25 17:22:16 -0400 (Thu, 25 May 2006) | 3 lines

 Oops.


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r30427 | file | 2006-05-25 16:30:40 -0500 (Thu, 25 May 2006) | 2 lines

Merge in branch that adds new features to MeetMeAdmin. See application documentation for more details on the new options. (issue #7131 reported by dmikusa branch by jcollie)

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r30430 | bweschke | 2006-05-25 16:47:03 -0500 (Thu, 25 May 2006) | 3 lines

  A new way to try and deal with deadlocks that occur in app_queue at present. Using this approach, we only manipulate the main queue mutexes when we get a dev state change on a device that is actually a member of a queue. Further optimizations are still possible (eg - store and manage pointers to the status integer of the member record that this interface/device has a one-to-one relationship with and then go directly to those pointers to make status modifications rather than the recursive looping that goes on now) BUT first things first. :)


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r30458 | russell | 2006-05-25 17:05:12 -0500 (Thu, 25 May 2006) | 2 lines

restore default paths for FreeBSD (reported by alphaque, fixed by jcollie)

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r30463 | russell | 2006-05-25 17:06:55 -0500 (Thu, 25 May 2006) | 2 lines

regenerate configure after the last fix

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r30465 | russell | 2006-05-25 17:39:57 -0500 (Thu, 25 May 2006) | 2 lines

only display a debug message if option_debug is in use

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r30490 | markster | 2006-05-25 22:08:15 -0500 (Thu, 25 May 2006) | 2 lines

Lets not commit things that cause Asterisk to break when config files aren't present.

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r30521 | markster | 2006-05-26 00:21:41 -0500 (Fri, 26 May 2006) | 2 lines

That goes for jingle too :)

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r30547 | file | 2006-05-26 12:43:11 -0500 (Fri, 26 May 2006) | 2 lines

Add the video stream for AGI function STREAM FILE (issue #5392 reported by areski -- minor mods by me)

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r30548 | file | 2006-05-26 12:59:29 -0500 (Fri, 26 May 2006) | 2 lines

attended transfer use transferer context first and set who is transfering at the beginning (issue #6752 reported by moy -- minor mods done by myself)

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r30578 | russell | 2006-05-26 13:19:37 -0500 (Fri, 26 May 2006) | 2 lines

add some more text about the build system

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r30579 | russell | 2006-05-26 13:25:38 -0500 (Fri, 26 May 2006) | 2 lines

wrap test at 80 characters

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r30580 | russell | 2006-05-26 13:33:58 -0500 (Fri, 26 May 2006) | 3 lines

document the changes I made yesterday to the exit behavior of the
AGI applications

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r30603 | file | 2006-05-26 14:48:17 -0500 (Fri, 26 May 2006) | 2 lines

Add ability to disable log / verbose output to remote consoles (issue #6524 reported by mavetju)

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r30607 | file | 2006-05-26 15:00:48 -0500 (Fri, 26 May 2006) | 2 lines

Few more expire_registry changes

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r30630 | russell | 2006-05-26 16:47:52 -0500 (Fri, 26 May 2006) | 17 lines

Fix various problems in the addition of the ability to mute log/verbose
output to remove consoles. The prototypes added to logger.h still need
doxygen documentation, as well.

- Add the new command line option to the man page
- make the mute option a flag instead of an int since it is only a binary
  option
- remove useless extern keywords for prototypes added to logger.h
- rename ast_console_mute() to ast_console_toggle_mute() since that is what
  it actually does
- actually apply the mute option to newly created remote consoles instead of
  only working when the CLI command is used
- don't imply the NO_FORK option if the mute command line option is provided
- place the new CLI command in the correct place in the list which has to be
  in alphabetical order
- Finally, clean up a few spacing issues to conform to the coding guidelines

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r30653 | tilghman | 2006-05-27 13:19:16 -0500 (Sat, 27 May 2006) | 2 lines

Should use the named handle, not one hardcoded

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r30654 | tilghman | 2006-05-27 13:45:15 -0500 (Sat, 27 May 2006) | 2 lines

Notate that QUEUEAGENTCOUNT is deprecated, so it can be removed post-1.4

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r30655 | russell | 2006-05-27 13:47:44 -0500 (Sat, 27 May 2006) | 1 line

make some variables static ... committed from xcode :)
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r30677 | tilghman | 2006-05-28 10:10:19 -0500 (Sun, 28 May 2006) | 2 lines

Deprecate SetGlobalVar, replacing it with a dialplan function

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r30699 | rizzo | 2006-05-29 00:13:13 -0500 (Mon, 29 May 2006) | 3 lines

this file contained the body twice, so remove the second instance.


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r30700 | rizzo | 2006-05-29 00:14:52 -0500 (Mon, 29 May 2006) | 4 lines

remove an explicit constant;
add a comment on the need to sort patterns in the standard way.


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r30701 | rizzo | 2006-05-29 00:23:59 -0500 (Mon, 29 May 2006) | 3 lines

remove unused include


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r30723 | russell | 2006-05-29 09:52:55 -0500 (Mon, 29 May 2006) | 2 lines

remove duplicate static keywords, oops

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r30746 | russell | 2006-05-30 06:23:48 -0500 (Tue, 30 May 2006) | 3 lines

remove a bunch of duplicated log messages.  There is a warning that gets
logged when this function returns an error

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r30771 | bweschke | 2006-05-30 09:59:02 -0500 (Tue, 30 May 2006) | 3 lines

 It's a 1.2 'thang'. 


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r30800 | kpfleming | 2006-05-30 11:01:50 -0500 (Tue, 30 May 2006) | 2 lines

fix various typos and other bits (from Ian Kinner)

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r30803 | kpfleming | 2006-05-30 11:08:38 -0500 (Tue, 30 May 2006) | 10 lines

Merged revisions 30802 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r30802 | kpfleming | 2006-05-30 11:07:16 -0500 (Tue, 30 May 2006) | 2 lines

another S/390 build fix

........

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r30804 | file | 2006-05-30 11:10:10 -0500 (Tue, 30 May 2006) | 2 lines

Be gone unused res! (issue #7238 reported by casper)

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r30805 | file | 2006-05-30 11:23:34 -0500 (Tue, 30 May 2006) | 2 lines

Be gone foul Makefile usage! er I mean use the correct variable... UTILS instead of TARGET. (issue #7239 reported by casper)

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r30806 | jcollie | 2006-05-30 11:36:21 -0500 (Tue, 30 May 2006) | 1 line

Get rid of warning about datarootdir at the end of the configure process...
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r30835 | file | 2006-05-30 12:33:37 -0500 (Tue, 30 May 2006) | 2 lines

Remove ourselves from the userlist a little bit earlier

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r30837 | kpfleming | 2006-05-30 12:52:49 -0500 (Tue, 30 May 2006) | 2 lines

officially deprecate the 'roundrobin' queue strategy in favor of 'rrmemory'

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r30847 | kpfleming | 2006-05-30 13:01:52 -0500 (Tue, 30 May 2006) | 2 lines

when we receive an IAX2 registration request with both a plaintext secret and an MD5 challenge, prefer the MD5 challenge for authentation (reported on asterisk-dev)

................
r30875 | kpfleming | 2006-05-30 14:20:20 -0500 (Tue, 30 May 2006) | 10 lines

Merged revisions 30874 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r30874 | kpfleming | 2006-05-30 14:18:30 -0500 (Tue, 30 May 2006) | 2 lines

check the proper variable...

........

................

Added:
    team/jcollie/bug7021/channels/chan_jingle.c
      - copied unchanged from r30875, trunk/channels/chan_jingle.c
    team/jcollie/bug7021/configs/jabber.conf.sample
      - copied unchanged from r30875, trunk/configs/jabber.conf.sample
    team/jcollie/bug7021/configs/jingle.conf.sample
      - copied unchanged from r30875, trunk/configs/jingle.conf.sample
    team/jcollie/bug7021/doc/callfiles.txt
      - copied unchanged from r30875, trunk/doc/callfiles.txt
    team/jcollie/bug7021/doc/jabber.txt
      - copied unchanged from r30875, trunk/doc/jabber.txt
    team/jcollie/bug7021/doc/jingle.txt
      - copied unchanged from r30875, trunk/doc/jingle.txt
    team/jcollie/bug7021/funcs/func_global.c
      - copied unchanged from r30875, trunk/funcs/func_global.c
    team/jcollie/bug7021/include/asterisk/jabber.h
      - copied unchanged from r30875, trunk/include/asterisk/jabber.h
    team/jcollie/bug7021/include/asterisk/jingle.h
      - copied unchanged from r30875, trunk/include/asterisk/jingle.h
    team/jcollie/bug7021/res/res_jabber.c
      - copied unchanged from r30875, trunk/res/res_jabber.c
Modified:
    team/jcollie/bug7021/   (props changed)
    team/jcollie/bug7021/CREDITS
    team/jcollie/bug7021/Makefile
    team/jcollie/bug7021/UPGRADE.txt
    team/jcollie/bug7021/apps/app_alarmreceiver.c
    team/jcollie/bug7021/apps/app_db.c
    team/jcollie/bug7021/apps/app_dial.c
    team/jcollie/bug7021/apps/app_directory.c
    team/jcollie/bug7021/apps/app_disa.c
    team/jcollie/bug7021/apps/app_echo.c
    team/jcollie/bug7021/apps/app_lookupblacklist.c
    team/jcollie/bug7021/apps/app_lookupcidname.c
    team/jcollie/bug7021/apps/app_meetme.c
    team/jcollie/bug7021/apps/app_playback.c
    team/jcollie/bug7021/apps/app_queue.c
    team/jcollie/bug7021/apps/app_rpt.c
    team/jcollie/bug7021/apps/app_setcallerid.c
    team/jcollie/bug7021/apps/app_setcdruserfield.c
    team/jcollie/bug7021/apps/app_settransfercapability.c
    team/jcollie/bug7021/apps/app_speech_utils.c
    team/jcollie/bug7021/apps/app_stack.c
    team/jcollie/bug7021/apps/app_waitforsilence.c
    team/jcollie/bug7021/asterisk.8
    team/jcollie/bug7021/asterisk.c
    team/jcollie/bug7021/build_tools/menuselect-deps.in
    team/jcollie/bug7021/cdr/cdr_radius.c
    team/jcollie/bug7021/channel.c
    team/jcollie/bug7021/channels/Makefile
    team/jcollie/bug7021/channels/chan_iax2.c
    team/jcollie/bug7021/channels/chan_misdn.c
    team/jcollie/bug7021/channels/chan_sip.c
    team/jcollie/bug7021/channels/chan_zap.c
    team/jcollie/bug7021/channels/misdn/chan_misdn_config.h
    team/jcollie/bug7021/channels/misdn/isdn_lib.c
    team/jcollie/bug7021/channels/misdn/isdn_lib.h
    team/jcollie/bug7021/channels/misdn/isdn_msg_parser.c
    team/jcollie/bug7021/channels/misdn_config.c
    team/jcollie/bug7021/cli.c
    team/jcollie/bug7021/codecs/codec_a_mu.c
    team/jcollie/bug7021/codecs/codec_ulaw.c
    team/jcollie/bug7021/codecs/gsm/Makefile
    team/jcollie/bug7021/configs/extensions.conf.sample
    team/jcollie/bug7021/configs/features.conf.sample
    team/jcollie/bug7021/configs/misdn.conf.sample
    team/jcollie/bug7021/configs/sip.conf.sample
    team/jcollie/bug7021/configs/voicemail.conf.sample
    team/jcollie/bug7021/configure
    team/jcollie/bug7021/configure.ac
    team/jcollie/bug7021/doc/channelvariables.txt
    team/jcollie/bug7021/funcs/func_channel.c
    team/jcollie/bug7021/funcs/func_db.c
    team/jcollie/bug7021/funcs/func_odbc.c
    team/jcollie/bug7021/include/asterisk/app.h
    team/jcollie/bug7021/include/asterisk/channel.h
    team/jcollie/bug7021/include/asterisk/devicestate.h
    team/jcollie/bug7021/include/asterisk/linkedlists.h
    team/jcollie/bug7021/include/asterisk/lock.h
    team/jcollie/bug7021/include/asterisk/logger.h
    team/jcollie/bug7021/include/asterisk/options.h
    team/jcollie/bug7021/include/autoconfig.h.in
    team/jcollie/bug7021/logger.c
    team/jcollie/bug7021/makeopts.in
    team/jcollie/bug7021/pbx.c
    team/jcollie/bug7021/pbx/Makefile
    team/jcollie/bug7021/pbx/pbx_spool.c
    team/jcollie/bug7021/res/Makefile
    team/jcollie/bug7021/res/res_agi.c
    team/jcollie/bug7021/res/res_features.c
    team/jcollie/bug7021/sample.call
    team/jcollie/bug7021/utils/Makefile

Propchange: team/jcollie/bug7021/
------------------------------------------------------------------------------
    automerge = *

Propchange: team/jcollie/bug7021/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.

Propchange: team/jcollie/bug7021/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/jcollie/bug7021/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue May 30 14:35:02 2006
@@ -1,1 +1,1 @@
-/trunk:1-29522
+/trunk:1-30875

Modified: team/jcollie/bug7021/CREDITS
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug7021/CREDITS?rev=30876&r1=30875&r2=30876&view=diff
==============================================================================
--- team/jcollie/bug7021/CREDITS (original)
+++ team/jcollie/bug7021/CREDITS Tue May 30 14:35:02 2006
@@ -79,6 +79,8 @@
 Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade
 Claude Patry - bug fixes, feature enhancements, and bug marshalling
 	cpatry at gmail.com
+Miroslav Nachev, miro at space-comm.com COSMOS Software Enterprises, Ltd.
+	- for Variable for No Answer Timeout for Attended Transfer
 
 === OTHER CONTRIBUTIONS ===
 John Todd - Monkey sounds and associated teletorture prompt

Modified: team/jcollie/bug7021/Makefile
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug7021/Makefile?rev=30876&r1=30875&r2=30876&view=diff
==============================================================================
--- team/jcollie/bug7021/Makefile (original)
+++ team/jcollie/bug7021/Makefile Tue May 30 14:35:02 2006
@@ -95,14 +95,18 @@
 else
   ASTETCDIR=$(sysconfdir)/asterisk
   ASTLIBDIR=$(libdir)/asterisk
-  ASTVARLIBDIR=$(localstatedir)/lib/asterisk
-  ASTSPOOLDIR=$(localstatedir)/spool/asterisk
-  ASTLOGDIR=$(localstatedir)/log/asterisk
   ASTHEADERDIR=$(includedir)/asterisk
   ASTBINDIR=$(bindir)
   ASTSBINDIR=$(sbindir)
+  ASTSPOOLDIR=$(localstatedir)/spool/asterisk
+  ASTLOGDIR=$(localstatedir)/log/asterisk
   ASTVARRUNDIR=$(localstatedir)/run
   ASTMANDIR=$(mandir)
+ifeq ($(OSARCH),FreeBSD)
+  ASTVARLIBDIR=$(prefix)/share/asterisk
+else
+  ASTVARLIBDIR=$(localstatedir)/lib/asterisk
+endif
 endif
 ASTDATADIR?=$(ASTVARLIBDIR)
 

Modified: team/jcollie/bug7021/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug7021/UPGRADE.txt?rev=30876&r1=30875&r2=30876&view=diff
==============================================================================
--- team/jcollie/bug7021/UPGRADE.txt (original)
+++ team/jcollie/bug7021/UPGRADE.txt Tue May 30 14:35:02 2006
@@ -14,30 +14,46 @@
 be present, etc.
 
 You must run the configure script before Asterisk will build, although it will
-attempt to automatically run it for you with no options specified; for most users,
-that will result in a similar build to what they would have had before the
-configure script was added to the build process (except for having to run 'make'
-again after the configure script is run). Note that the configure script does NOT
-need to be re-run just to rebuild Asterisk; you only need to re-run it when your
-system configuration changes or you wish to build Asterisk with different options.
+attempt to automatically run it for you with no options specified; for most
+users, that will result in a similar build to what they would have had before
+the configure script was added to the build process (except for having to run
+'make' again after the configure script is run). Note that the configure script
+does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
+when your system configuration changes or you wish to build Asterisk with 
+different options.
 
 Build Process (module selection):
 
 The Asterisk source tree now includes a basic module selection and build option
 selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
 In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not (and cannot)
-be built because your system does not have the required external dependencies
-installed.
-
-(TODO: document where 'global' and 'per-user' menuselect input files should go
-and what they need to contain)
+turn on/off global options for the build and see which modules will not 
+(and cannot) be built because your system does not have the required external
+dependencies installed.
+
+The resulting file from menuselect is called 'menuselect.makeopts'. Note that
+the resulting menuselect.makeopts file generally contains which modules *not*
+to build. The modules listed in this file indicate which modules have unmet
+dependencies, a present conflict, or have been disabled by the user in the
+menuselect interface. Compiler Flags can also be set in the menuselect
+interface.  In this case, the resulting file contains which CFLAGS are in use,
+not which ones are not in use.
+
+If you would like to save your choices and have them applied against all
+builds, the file can be copied to '~/.asterisk.makeopts' or 
+'/etc/asterisk.makeopts'.
 
 PBX Core:
 
 * The (very old and undocumented) ability to use BYEXTENSION for dialing
   instead of ${EXTEN} has been removed.
-
+  
+* Builtin (res_features) transfer functionality attempts to use the context
+  defined in TRANSFER_CONTEXT variable of the transferer channel first. If
+  not set, it uses the transferee variable. If not set in any channel, it will 
+  attempt to use the last non macro context. If not possible, it will default
+  to the current context.
+ 
 Command Line Interface:
 
 * 'show channels concise', designed to be used by applications that will parse
@@ -58,6 +74,12 @@
   and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
   been removed in this version.  You should use the equivalent dialplan
   function in places where you have previously used one of these applications.
+
+* The application SetGlobalVar has been deprecated.  You should replace uses
+  of this application with the following combination of Set and GLOBAL():
+  Set(GLOBAL(name)=value).  You may also access global variables exclusively by
+  using the GLOBAL() dialplan function, instead of relying on variable
+  interpolation falling back to globals when no channel variable is set.
 
 * The application SetVar has been renamed to Set.  The syntax SetVar was marked
   deprecated in version 1.2 and is no longer recognized in this version.
@@ -97,6 +119,17 @@
   record conversations queue members are having with queue callers. Please
   see configs/queues.conf.sample for more information on this option.
 
+* The app_queue application strategy called 'roundrobin' has been deprecated
+  for this release. Users are encouraged to use 'rrmemory' instead, since it
+  provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
+  'rrmemory' will be renamed 'roundrobin'.
+
+* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
+  the 'm' option now provides the functionality of "initially muted". 
+  In practice, most existing dialplans using the 'm' flag should not notice
+  any difference, unless the keypad menu is enabled, allowing the user 
+  to unmute themsleves.
+
 * ast_play_and_record would attempt to cancel the recording if a DTMF
   '0' was received.  This behavior was not documented in most of the
   applications that used ast_play_and_record and the return codes from
@@ -113,6 +146,24 @@
   This addresses the deficiency of not being able to count the number of
   messages in folders other than INBOX and Old.
 
+* The exit behavior of the AGI applications has changed. Previously, when
+  a connection to an AGI server failed, the application would cause the channel
+  to immediately stop dialplan execution and hangup. Now, the only time that
+  the AGI applications will cause the channel to stop dialplan execution is
+  when the channel itself requests hangup. The AGI applications now set an
+  AGISTATUS variable which will allow you to find out whether running the AGI
+  was successful or not.
+
+  Previously, there was no way to handle the case where Asterisk was unable to
+  locally execute an AGI script for some reason. In this case, dialplan
+  execution will continue as it did before, but the AGISTATUS variable will be
+  set to "FAILURE".
+
+  A locally executed AGI script can now exit with a non-zero exit code and this
+  failure will be detected by Asterisk. If an AGI script exits with a non-zero
+  exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
+  "SUCCESS".
+
 Manager:
 
 * After executing the 'status' manager action, the "Status" manager events
@@ -132,6 +183,12 @@
   headers are not automatically sent, unless you specify them as separate
   arguments.  Please see the application help for the new syntax.
 
+* app_meetme: Mute and Unmute events are now reported via the Manager API.
+  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
+  are easier to use than "Action Command:". The MeetMeStopTalking event has
+  also been deprecated in favor of the already existing MeetmeTalking event
+  with a "Status" of "on" or "off" added.
+
 Variables:
 
 * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
@@ -146,6 +203,9 @@
 * OSP applications exports several new variables, ${OSPINHANDLE},
   ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
   ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
+  
+* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
+  created channel. This variables holds the channel name of the transferer.
 
 Functions:
 
@@ -157,13 +217,17 @@
   modules.conf file then you will need to explicitly load the modules that
   contain the functions you want to use.
 
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of records),
-  but the lookup fails to match any records, the returned value will now be "0" instead of blank.
+* The ENUMLOOKUP() function with the 'c' option (for counting the number of 
+  records), but the lookup fails to match any records, the returned value will 
+  now be "0" instead of blank.
 
 * The REALTIME() function is now available in version 1.4 and app_realtime has
   been deprecated in favor of the new function. app_realtime will be removed
   completely with the version 1.6 release so please take the time between
   releases to make any necessary changes
+
+* The QUEUEAGENTCOUNT() function has been deprecated in favor of
+  QUEUE_MEMBER_COUNT().
 
 The IAX2 channel:
 
@@ -176,7 +240,8 @@
 
 The SIP channel:
 

[... 27892 lines stripped ...]