[asterisk-commits] branch jcollie/bug6082 r30545 - in /team/jcollie/bug6082: ./ apps/ build_tool...

[asterisk-commits at lists.digium.com]

Author: jcollie
Date: Fri May 26 10:30:52 2006
New Revision: 30545

URL: http://svn.digium.com/view/asterisk?rev=30545&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 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
................
r29554 | kpfleming | 2006-05-22 16:20:01 -0500 (Mon, 22 May 2006) | 2 lines

bootstrap updates to include xmpp related stuff

................
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)

........

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

remove properties from the xmpp merge

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

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

Handle ringing (early) state properly on SIP

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

Formatting, typos

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

Code formatting

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


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

Breaking once will stop us... :-)

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

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

................
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)


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

........
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)


........

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

remove another excess "debug" message

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

................
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)

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

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


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


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

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

simplify/fix lock retry, and fix comment

........

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

................
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.
................
r29825 | mogorman | 2006-05-23 15:25:23 -0500 (Tue, 23 May 2006) | 2 lines

fixes bug where server goes away.

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

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

fix the sourceaddress option (issue #7213, alphaque)

........

................
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)

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

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

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

........
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)

........

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

........
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)

........

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

........
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)

........

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

Typo fix. Thanks Peter!

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

........
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)

........

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

Blocked revisions 30035 via svnmerge

........
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

........

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

Typo

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

reset timeout on reconnect.

................
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)

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

........
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

........

................
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)

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

block chan_sip fix that is not broken here :-)

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

Update some documentation (file internal brain bug #42)

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

Nothing to see here... move along

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


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

 Doxygen comment for Qwell


................
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! 


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


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

........
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)

........

................
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)

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

................
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)

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

........
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)

........

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

........
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)

........

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

................
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)

................
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)

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

........
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)

........

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

........
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. :)


................
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)

................
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 :)

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

Added:
    team/jcollie/bug6082/channels/chan_jingle.c
      - copied unchanged from r30521, trunk/channels/chan_jingle.c
    team/jcollie/bug6082/configs/jabber.conf.sample
      - copied unchanged from r30521, trunk/configs/jabber.conf.sample
    team/jcollie/bug6082/configs/jingle.conf.sample
      - copied unchanged from r30521, trunk/configs/jingle.conf.sample
    team/jcollie/bug6082/doc/callfiles.txt
      - copied unchanged from r30521, trunk/doc/callfiles.txt
    team/jcollie/bug6082/doc/jabber.txt
      - copied unchanged from r30521, trunk/doc/jabber.txt
    team/jcollie/bug6082/doc/jingle.txt
      - copied unchanged from r30521, trunk/doc/jingle.txt
    team/jcollie/bug6082/include/asterisk/jabber.h
      - copied unchanged from r30521, trunk/include/asterisk/jabber.h
    team/jcollie/bug6082/include/asterisk/jingle.h
      - copied unchanged from r30521, trunk/include/asterisk/jingle.h
    team/jcollie/bug6082/res/res_jabber.c
      - copied unchanged from r30521, trunk/res/res_jabber.c
Modified:
    team/jcollie/bug6082/   (props changed)
    team/jcollie/bug6082/CREDITS
    team/jcollie/bug6082/Makefile
    team/jcollie/bug6082/UPGRADE.txt
    team/jcollie/bug6082/apps/app_db.c
    team/jcollie/bug6082/apps/app_dial.c
    team/jcollie/bug6082/apps/app_echo.c
    team/jcollie/bug6082/apps/app_lookupblacklist.c
    team/jcollie/bug6082/apps/app_lookupcidname.c
    team/jcollie/bug6082/apps/app_meetme.c
    team/jcollie/bug6082/apps/app_queue.c
    team/jcollie/bug6082/apps/app_rpt.c
    team/jcollie/bug6082/apps/app_setcallerid.c
    team/jcollie/bug6082/apps/app_setcdruserfield.c
    team/jcollie/bug6082/apps/app_settransfercapability.c
    team/jcollie/bug6082/apps/app_speech_utils.c
    team/jcollie/bug6082/apps/app_waitforsilence.c
    team/jcollie/bug6082/asterisk.c
    team/jcollie/bug6082/build_tools/menuselect-deps.in
    team/jcollie/bug6082/channel.c
    team/jcollie/bug6082/channels/Makefile
    team/jcollie/bug6082/channels/chan_iax2.c
    team/jcollie/bug6082/channels/chan_misdn.c
    team/jcollie/bug6082/channels/chan_sip.c
    team/jcollie/bug6082/channels/chan_zap.c
    team/jcollie/bug6082/channels/misdn/chan_misdn_config.h
    team/jcollie/bug6082/channels/misdn/isdn_lib.c
    team/jcollie/bug6082/channels/misdn/isdn_lib.h
    team/jcollie/bug6082/channels/misdn/isdn_msg_parser.c
    team/jcollie/bug6082/channels/misdn_config.c
    team/jcollie/bug6082/codecs/codec_a_mu.c
    team/jcollie/bug6082/codecs/codec_ulaw.c
    team/jcollie/bug6082/codecs/gsm/Makefile
    team/jcollie/bug6082/configs/extensions.conf.sample
    team/jcollie/bug6082/configs/features.conf.sample
    team/jcollie/bug6082/configs/misdn.conf.sample
    team/jcollie/bug6082/configs/sip.conf.sample
    team/jcollie/bug6082/configure
    team/jcollie/bug6082/configure.ac
    team/jcollie/bug6082/doc/channelvariables.txt
    team/jcollie/bug6082/funcs/func_channel.c
    team/jcollie/bug6082/funcs/func_db.c
    team/jcollie/bug6082/funcs/func_odbc.c
    team/jcollie/bug6082/include/asterisk/app.h
    team/jcollie/bug6082/include/asterisk/channel.h
    team/jcollie/bug6082/include/asterisk/devicestate.h
    team/jcollie/bug6082/include/asterisk/linkedlists.h
    team/jcollie/bug6082/include/asterisk/lock.h
    team/jcollie/bug6082/include/autoconfig.h.in
    team/jcollie/bug6082/logger.c
    team/jcollie/bug6082/makeopts.in
    team/jcollie/bug6082/pbx.c
    team/jcollie/bug6082/pbx/Makefile
    team/jcollie/bug6082/pbx/pbx_spool.c
    team/jcollie/bug6082/res/Makefile
    team/jcollie/bug6082/res/res_agi.c
    team/jcollie/bug6082/res/res_features.c
    team/jcollie/bug6082/sample.call
    team/jcollie/bug6082/utils/Makefile

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------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri May 26 10:30:52 2006
@@ -1,1 +1,1 @@
-/trunk:1-29521
+/trunk:1-30544

Modified: team/jcollie/bug6082/CREDITS
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/CREDITS?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/CREDITS (original)
+++ team/jcollie/bug6082/CREDITS Fri May 26 10:30:52 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/bug6082/Makefile
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/Makefile?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/Makefile (original)
+++ team/jcollie/bug6082/Makefile Fri May 26 10:30:52 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/bug6082/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/UPGRADE.txt?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/UPGRADE.txt (original)
+++ team/jcollie/bug6082/UPGRADE.txt Fri May 26 10:30:52 2006
@@ -97,6 +97,12 @@
   record conversations queue members are having with queue callers. Please
   see configs/queues.conf.sample for more information on this option.
 
+* 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
@@ -132,6 +138,12 @@
   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},

Modified: team/jcollie/bug6082/apps/app_db.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_db.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_db.c (original)
+++ team/jcollie/bug6082/apps/app_db.c Fri May 26 10:30:52 2006
@@ -48,9 +48,11 @@
 #include "asterisk/lock.h"
 #include "asterisk/options.h"
 
+/*! \todo XXX Remove this application after 1.4 is relased */
 static char *d_descrip =
 "  DBdel(family/key): This applicaiton will delete a key from the Asterisk\n"
-"database.\n";
+"database.\n"
+"  This application has been DEPRECATED in favor of the DB_DELETE function.\n";
 
 static char *dt_descrip =
 "  DBdeltree(family[/keytree]): This application will delete a family or keytree\n"
@@ -109,8 +111,14 @@
 {
 	char *argv, *family, *key;
 	struct localuser *u;
+	static int deprecation_warning = 0;
 
 	LOCAL_USER_ADD(u);
+
+	if (!deprecation_warning) {
+		deprecation_warning = 1;
+		ast_log(LOG_WARNING, "The DBdel application has been deprecated in favor of the DB_DELETE dialplan function!\n");
+	}
 
 	argv = ast_strdupa(data);
 

Modified: team/jcollie/bug6082/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_dial.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_dial.c (original)
+++ team/jcollie/bug6082/apps/app_dial.c Fri May 26 10:30:52 2006
@@ -959,21 +959,27 @@
 		}
 		
 		if(privdb_val == AST_PRIVACY_DENY ) {
+			ast_copy_string(status, "NOANSWER", sizeof(status));
 			if (option_verbose > 2)
 				ast_verbose( VERBOSE_PREFIX_3  "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
 			res=0;
 			goto out;
 		}
 		else if(privdb_val == AST_PRIVACY_KILL ) {
-			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
+			ast_copy_string(status, "DONTCALL", sizeof(status));
+			if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
+				ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
+			}
 			res = 0;
 			goto out; /* Is this right? */
 		}
 		else if(privdb_val == AST_PRIVACY_TORTURE ) {
-			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
+			ast_copy_string(status, "TORTURE", sizeof(status));
+			if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
+				ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
+			}
 			res = 0;
 			goto out; /* is this right??? */
-
 		}
 		else if(privdb_val == AST_PRIVACY_UNKNOWN ) {
 			/* Get the user's intro, store it in priv-callerintros/$CID, 
@@ -1000,6 +1006,8 @@
 				ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0);  /* NOTE: I've reduced the total time to 4 sec */
 										/* don't think we'll need a lock removed, we took care of
 										   conflicts by naming the privintro file */
+				if( !ast_streamfile(chan, "vm-dialout", chan->language) )
+					ast_waitstream(chan, "");
 			}
 		}
 	}
@@ -1312,6 +1320,7 @@
 							     opt_args[OPT_ARG_PRIVACY], privcid);
 					ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_DENY);
 				}
+				ast_copy_string(status, "NOANSWER", sizeof(status));
 				ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
 				res=0;
 				goto out;

Modified: team/jcollie/bug6082/apps/app_echo.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_echo.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_echo.c (original)
+++ team/jcollie/bug6082/apps/app_echo.c Fri May 26 10:30:52 2006
@@ -72,26 +72,30 @@
 			break;
 		f->delivery.tv_sec = 0;
 		f->delivery.tv_usec = 0;
-		if (f->frametype == AST_FRAME_VOICE) {
-			if (ast_write(chan, f)) 
-				break;
-		} else if (f->frametype == AST_FRAME_VIDEO) {
-			if (ast_write(chan, f)) 
-				break;
-		} else if (f->frametype == AST_FRAME_DTMF) {
+		switch (f->frametype) {
+		case AST_FRAME_DTMF:
+		case AST_FRAME_DTMF_END:
 			if (f->subclass == '#') {
 				res = 0;
-				break;
-			} else {
-				if (ast_write(chan, f))
-					break;
+				ast_frfree(f);
+				goto end;
+			}
+			/* fall through */
+		case AST_FRAME_DTMF_BEGIN:
+		case AST_FRAME_VOICE:
+		case AST_FRAME_VIDEO:
+		case AST_FRAME_TEXT:
+		case AST_FRAME_HTML:
+		case AST_FRAME_IMAGE:
+			if (ast_write(chan, f)) {
+				ast_frfree(f);
+				goto end;
 			}
 		}
 		ast_frfree(f);
 	}
-
+end:
 	LOCAL_USER_REMOVE(u);
-
 	return res;
 }
 

Modified: team/jcollie/bug6082/apps/app_lookupblacklist.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_lookupblacklist.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_lookupblacklist.c (original)
+++ team/jcollie/bug6082/apps/app_lookupblacklist.c Fri May 26 10:30:52 2006
@@ -65,6 +65,32 @@
 
 LOCAL_USER_DECL;
 
+static int blacklist_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	char blacklist[1];
+	int bl = 0;
+
+	if (chan->cid.cid_num) {
+		if (!ast_db_get("blacklist", chan->cid.cid_num, blacklist, sizeof (blacklist)))
+			bl = 1;
+	}
+	if (chan->cid.cid_name) {
+		if (!ast_db_get("blacklist", chan->cid.cid_name, blacklist, sizeof (blacklist)))
+			bl = 1;
+	}
+
+	snprintf(buf, len, "%d", bl);
+	return 0;
+}
+
+static struct ast_custom_function blacklist_function = {
+	.name = "BLACKLIST",
+	.synopsis = "Check if the callerid is on the blacklist",
+	.desc = "Uses astdb to check if the Caller*ID is in family 'blacklist'.  Returns 1 or 0.\n",
+	.syntax = "BLACKLIST()",
+	.read = blacklist_read,
+};
+
 static int
 lookupblacklist_exec (struct ast_channel *chan, void *data)
 {
@@ -72,8 +98,14 @@
 	struct localuser *u;
 	int bl = 0;
 	int priority_jump = 0;
+	static int dep_warning = 0;
 
 	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		dep_warning = 1;
+		ast_log(LOG_WARNING, "LookupBlacklist is deprecated.  Please use ${BLACKLIST()} instead.\n");
+	}
 
 	if (!ast_strlen_zero(data)) {
 		if (strchr(data, 'j'))
@@ -112,6 +144,7 @@
 	int res;
 
 	res = ast_unregister_application(app);
+	res |= ast_custom_function_unregister(&blacklist_function);
 
 	STANDARD_HANGUP_LOCALUSERS;
 
@@ -120,7 +153,9 @@
 
 static int load_module(void *mod)
 {
-	return ast_register_application (app, lookupblacklist_exec, synopsis,descrip);
+	int res = ast_custom_function_register(&blacklist_function);
+	res |= ast_register_application (app, lookupblacklist_exec, synopsis,descrip);
+	return res;
 }
 
 static const char *description(void)

Modified: team/jcollie/bug6082/apps/app_lookupcidname.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_lookupcidname.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_lookupcidname.c (original)
+++ team/jcollie/bug6082/apps/app_lookupcidname.c Fri May 26 10:30:52 2006
@@ -66,8 +66,13 @@
 {
   char dbname[64];
   struct localuser *u;
+	static int dep_warning = 0;
 
   LOCAL_USER_ADD (u);
+	if (!dep_warning) {
+		dep_warning = 1;
+		ast_log(LOG_WARNING, "LookupCIDName is deprecated.  Please use ${DB(cidname/${CALLERID(num)})} instead.\n");
+	}
   if (chan->cid.cid_num) {
 	if (!ast_db_get ("cidname", chan->cid.cid_num, dbname, sizeof (dbname))) {
 		ast_set_callerid (chan, NULL, dbname, NULL);

Modified: team/jcollie/bug6082/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/team/jcollie/bug6082/apps/app_meetme.c?rev=30545&r1=30544&r2=30545&view=diff
==============================================================================
--- team/jcollie/bug6082/apps/app_meetme.c (original)
+++ team/jcollie/bug6082/apps/app_meetme.c Fri May 26 10:30:52 2006
@@ -74,8 +74,9 @@
 #define DEFAULT_AUDIO_BUFFERS  32
 
 enum {
-	ADMINFLAG_MUTED =  (1 << 1), /*!< User is muted */
-	ADMINFLAG_KICKME = (1 << 2)  /*!< User has been kicked */
+	ADMINFLAG_MUTED =     (1 << 1), /*!< User is muted */
+	ADMINFLAG_SELFMUTED = (1 << 2), /*!< User muted self */
+	ADMINFLAG_KICKME =    (1 << 3)  /*!< User has been kicked */
 };
 
 #define MEETME_DELAYDETECTTALK     300
@@ -146,9 +147,11 @@
 	/*! If set, won't speak the extra prompt when the first person 
 	 *  enters the conference */
 	CONFFLAG_NOONLYPERSON = (1 << 22),
-	CONFFLAG_INTROUSERNOREVIEW = (1 << 23)
+	CONFFLAG_INTROUSERNOREVIEW = (1 << 23),
 	/*! If set, user will be asked to record name on entry of conference 
 	 *  without review */
+	CONFFLAG_STARTMUTED = (1 << 24)
+	/*! If set, the user will be initially muted by admin */
 };
 
 AST_APP_OPTIONS(meetme_opts, {
@@ -163,7 +166,7 @@
 	AST_APP_OPTION('i', CONFFLAG_INTROUSER ),
 	AST_APP_OPTION('I', CONFFLAG_INTROUSERNOREVIEW ),
 	AST_APP_OPTION('M', CONFFLAG_MOH ),
-	AST_APP_OPTION('m', CONFFLAG_MONITOR ),
+	AST_APP_OPTION('m', CONFFLAG_STARTMUTED ),
 	AST_APP_OPTION('o', CONFFLAG_OPTIMIZETALKER ),
 	AST_APP_OPTION('P', CONFFLAG_ALWAYSPROMPT ),
 	AST_APP_OPTION('p', CONFFLAG_POUNDEXIT ),
@@ -171,6 +174,7 @@
 	AST_APP_OPTION('r', CONFFLAG_RECORDCONF ),
 	AST_APP_OPTION('s', CONFFLAG_STARMENU ),
 	AST_APP_OPTION('T', CONFFLAG_MONITORTALKER ),
+	AST_APP_OPTION('l', CONFFLAG_MONITOR ),
 	AST_APP_OPTION('t', CONFFLAG_TALKER ),
 	AST_APP_OPTION('w', CONFFLAG_WAITMARKED ),
 	AST_APP_OPTION('X', CONFFLAG_EXIT_CONTEXT ),
@@ -207,7 +211,8 @@
 "      'E' -- select an empty pinless conference\n"
 "      'i' -- announce user join/leave with review\n"
 "      'I' -- announce user join/leave without review\n"
-"      'm' -- set monitor only mode (Listen only, no talking)\n"
+"      'l' -- set listen only mode (Listen only, no talking)\n"
+"      'm' -- set initially muted by admin\n"
 "      'M' -- enable music on hold when the conference has a single caller\n"
 "      'o' -- set talker optimization - treats talkers who aren't speaking as\n"
 "             being muted, meaning (a) No encode is done on transmission and\n"
@@ -249,6 +254,16 @@
 "      'M' -- Mute conference\n"
 "      'n' -- Unmute entire conference (except admin)\n"
 "      'N' -- Mute entire conference (except admin)\n"
+"      'r' -- Reset one user's volume settings\n"
+"      'R' -- Reset all users volume settings\n"
+"      's' -- Lower entire conference speaking volume\n"
+"      'S' -- Raise entire conference speaking volume\n"
+"      't' -- Lower one user's talk volume\n"
+"      'T' -- Lower all users talk volume\n"
+"      'u' -- Lower one user's listen volume\n"
+"      'U' -- Lower all users listen volume\n"
+"      'v' -- Lower entire conference listening volume\n"
+"      'V' -- Raise entire conference listening volume\n"
 "";
 
 struct ast_conference {
@@ -294,7 +309,7 @@
 	int talking;                            /*!< Is user talking */
 	int zapchannel;                         /*!< Is a Zaptel channel */
 	char usrvalue[50];                      /*!< Custom User Value */
-	char namerecloc[AST_MAX_EXTENSION];     /*!< Name Recorded file Location */
+	char namerecloc[PATH_MAX];		/*!< Name Recorded file Location */
 	time_t jointime;                        /*!< Time the user joined the conference */
 	struct volume talk;
 	struct volume listen;
@@ -878,6 +893,7 @@
 	int using_pseudo = 0;
 	int duration=20;
 	int hr, min, sec;
+	int sent_event = 0;
 	time_t now;
 	struct ast_dsp *dsp=NULL;
 	struct ast_app *app;
@@ -948,7 +964,7 @@
 
 	user->chan = chan;
 	user->userflags = confflags;
-	user->adminflags = 0;
+	user->adminflags = (confflags & CONFFLAG_STARTMUTED) ? ADMINFLAG_MUTED : 0;
 	user->talking = -1;
 	conf->users++;
 	/* Update table */
@@ -1130,17 +1146,20 @@
 	}
 	ast_log(LOG_DEBUG, "Placed channel %s in ZAP conf %d\n", chan->name, conf->zapconf);
 
-	manager_event(EVENT_FLAG_CALL, "MeetmeJoin", 
-		      "Channel: %s\r\n"
-		      "Uniqueid: %s\r\n"
-		      "Meetme: %s\r\n"
-		      "Usernum: %d\r\n",
-		      chan->name, chan->uniqueid, conf->confno, user->user_no);
+	if (!sent_event) {
+		manager_event(EVENT_FLAG_CALL, "MeetmeJoin", 
+			      "Channel: %s\r\n"
+			      "Uniqueid: %s\r\n"
+			      "Meetme: %s\r\n"
+			      "Usernum: %d\r\n",
+			      chan->name, chan->uniqueid, conf->confno, user->user_no);
+		sent_event = 1;
+	}
 
 	if (!firstpass && !(confflags & CONFFLAG_MONITOR) && !(confflags & CONFFLAG_ADMIN)) {
 		firstpass = 1;
 		if (!(confflags & CONFFLAG_QUIET))
-			if (!(confflags & CONFFLAG_WAITMARKED) || (conf->markedusers >= 1))
+			if (!(confflags & CONFFLAG_WAITMARKED) || ((confflags & CONFFLAG_MARKEDUSER) && (conf->markedusers >= 1)))
 				conf_play(chan, conf, ENTER);
 	}
 
@@ -1298,39 +1317,51 @@
 				break;
 			}
 	
-			/* Check if the admin changed my modes */
-			if (user->adminflags) {			
-				/* Set the new modes */
-				if ((user->adminflags & ADMINFLAG_MUTED) && (ztc.confmode & ZT_CONF_TALKER)) {

[... 26501 lines stripped ...]