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(issue #7077, patch by adomjan)


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r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines

Merge changes from team/russell/inband_dtmf ...

Fix some issues related to generating inband DTMF.  There are two changes here:

1)   The list of DTMF tones in the senddigit_begin() function explicitly
   specified 100ms of the tone followed by 100ms of silence.  This really
   broke things with the way that Asterisk now wants complete control
   over when the digit begins and ends.  So, regardless of what Asterisk
   really wanted to do, this was going to play out the tone at the length it
   wanted to.  This caused various problems like DTMF translation to inband to
   be extremely unreliable.
     The list of tones has been changed so that the correct DTMF tone is played
   indefinitely until Asterisk tells it to stop.

2) ast_write() had to be modified to let a DTMF_END frame get processed even
   when a generator is present.  This is how the tone will finally get stopped.

(issues #8944, #9250, #9348, maybe others.  Thanks to mdu113 from #8944 for
 the testing and feedback!)

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r62738 | murf | 2007-05-02 14:46:07 -0600 (Wed, 02 May 2007) | 9 lines

Merged revisions 62737 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1 line

Some tweaks to satisfy CDR bug 8796, where being in 'h' extension louses up the dst field
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r62739 | russell | 2007-05-02 15:55:00 -0500 (Wed, 02 May 2007) | 3 lines

Backport the change that only went in to trunk that fixes the command manager
action over http.  (reported internally by pari and bkruse)

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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines

Merged revisions 62691 via svnmerge from 
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines

Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.

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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line

a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048, patch from tim_ringenbach)


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not properly added.  Thanks to IgorG for pointing it out!  :)


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dead period with a non-responsive CLI after I issue "load chan_sip.so"


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But first, some serious SIP testing :-)


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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines

Don't unlock a channel that we already know does not exist (propably isue 8228)

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add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel.  You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)


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r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines

Merged revisions 62547 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines

Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful.  (issue #9338, patch by svanlund)

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r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007) | 2 lines

Bug 9590 - Memory leaks around find_user() (found by rayjay, different fixes by me)

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r62497 | russell | 2007-05-01 11:26:48 -0500 (Tue, 01 May 2007) | 11 lines

Merged revisions 62496 via svnmerge from 
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r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines

Add indications.conf information for the Philippines.
(issue #9525, reported and patched by loloski)

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entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
 modified and updated to trunk by me)


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file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)


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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines

Merged revisions 62417 via svnmerge from 
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines

This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)

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for a SIP channel.
(issue #9619, reported by jtodd, original patch by Corydon76, committed patch
 slightly modified by me)


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r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | 4 lines

When serving dynamic content, include a Cache-Control header to instruct the
browsers to not store the resulting content.  
(issue #9621, reported by Pari, patch by me)

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r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr 2007) | 2 lines

Remove unused (and potentially confusing) jitterbuffer options from sample config.

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r62369 | file | 2007-04-30 11:36:11 -0300 (Mon, 30 Apr 2007) | 10 lines

Merged revisions 62368 via svnmerge from 
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r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 lines

Update copyright notice. It's now the year 2007!

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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines

Fix a bug that made the "language" setting in zapata.conf not
functional.  (issue #9626, reported and fixed by sergee)

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r62299 | russell | 2007-04-28 16:56:20 -0500 (Sat, 28 Apr 2007) | 2 lines

Note that the "talker optimization" option will be enabled by default in 1.6

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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.

This set of changes introduces the first use of the API, as well.  I have
restructured the way that MWI (message waiting indication) is handled.  It is
now event based instead of polling based.  For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes.  app_voicemail will generate events
when changes occur.

See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective.  For developer information, see the text in
include/asterisk/event.h.

As always, additional feedback is welcome on the asterisk-dev mailing list.


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and DUNDIRESULT.  Also, update the automatically generated application docs.


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This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan.  Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.


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minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)


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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines

Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks.  This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel.  However, this is the
wrong thing to do.  It should *always* return 0, instead.  When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)

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r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, 27 Apr 2007) | 11 lines

Merged revisions 62173 via svnmerge from 
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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines

This transcoder message needn't be a NOTICE.
I've seen it cause confusion more than a few times.

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r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines

If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a 
channel.  Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)

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