This replaces tab completion code with the use of a public function that
does the same thing


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closes issue #11294; missed the conditional unlock of the contexts when the hash table is used instead; also, used the ast_free_ptr as advised.

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r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines

Merge changes from team/russell/autoservice_1.4

This set of changes fixes an issue that was reported to me on IRC yesterday.
The user, d1mas, was using chan_zap for incoming calls and was having DTMF
recognition issues in some situations.  Specifically, he noticed that the 
problem occurred when using DISA or WaitExten.  He also noticed that when 
using Read, the problem did not occur.  His system also used DUNDi for 
dialplan lookups.

So, he theorized that if the DUNDi lookups blocked for some period of time,
that audio from the zap channel could get lost.  If the audio got lost, then
it wouldn't be run through the DTMF detector, and digits could get lost.
He was correct, and the following set of changes fixes the problem.  However,
the changes go a little bit further than what was necessary to fix this exact
problem.

1) I updated pbx_extension_helper() to autoservice the associated channel to
   handle cases where extension lookups may take a long time.  This would
   normally be a dialplan switch that does some lookup over the network, such
   as the DUNDi or IAX2 switches.

   This ensures that even while a DUNDi lookup is blocking, the channel will be
   continuously serviced.

2) I made a change to the autoservice code.  This is actually something that
   has bothered me for a long time.  When a channel is in autoservice, _all_
   frames get thrown away.  However, some frames really shouldn't be thrown
   away.  The most notable examples are signalling (CONTROL) frames, and DTMF.

   So, this patch queues up important frames while a channel is in autoservice.
   When autoservice is stopped on the channel, the queued up frames get stuck
   back on the channel so that they can get processed instead of thrown away.

3) I made another change to the autoservice code to handle the case where
   autoservice is started on channels recursively.

   Previously, you could call ast_autoservice_start() multiple times on a
   channel, and it would stop the first time ast_autoservice_stop() gets
   called.  Now, it will ensure that autoservice doesn't actually stop until
   the final call to ast_autoservice_stop().

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before making more dramatic changes.


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r89727 | mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6 lines

Changing some calls from free() to ast_free() since they were allocated with
ast_calloc().

(closes issue #11390, reported and patched by Laureano)


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r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines

on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL

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r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines

generate a warning when an application option that requires an argument is ignored due to lack of an argument

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follow.


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Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...

Merged revisions 89630 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines

If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.

With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.

(closes issue #11376)
Reported by: lasse
Patches: 
      bug11376.txt uploaded by oej (license 306)
Tested by: lasse

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 except with an additional boolean arg.

A hack such as:
foo ? S_OR(bar, "baz") : "baz"
becomes:
S_COR(foo, bar, "baz")


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made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also.

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(closes issue #11348)
Reported by: sperreault


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r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines

Add a note to the sample voicemail config noting that when using IMAP storage,
only the first format specified will be attached to the message.

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r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines

Default result of STAT should be "0" not "".
Reported via the -users mailing list, fixed by me.

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r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines

Clarify limitonpeers=yes

(closes issue #11304)
Reported by: pj


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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line

closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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consistency's sake

(closes issue #11381, reported and patched by jon)



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r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines

After issuing a "say load new", if a caller hangs up during the middle of playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.

(closes issue #11345, reported and patched by IgorG)


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r89616 | mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5 lines

After issuing a "say load new" tons of warning messages are printed
out to the CLI every time do_say in app_playback is called. Removing these
warnings


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that was just added.

Cresl1n, please keep this in mind when making these changes to libpri or libss7.


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Both still works in this version.


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r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines

Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252)

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- Restructure other changes to UPGRADE.txt and CHANGES

We're still looking for scripts that replace 
	asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?


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Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.


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r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines

Add module counting removal for error conditions.
(closes issue #11333)
Reported by: Laureano
Patches:
      res_features_v2.c.patch uploaded by Laureano (license 265)

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r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines

Add channel locking to a function that needed to be doing it.  This is just a
little something I noticed while working on a completely unrelated issue.

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r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 lines

Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled.
(closes issue #11347)
Reported by: ys
Patches:
      pbx.pbx_config.c.diff uploaded by ys (license 281)

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closes issue #11356; Many thanks to snuffy for his code review and changes to cut down duplication. I tested this against hashtest, and it passes. I reviewed the changes, and they look reasonable. I had to remove a few const decls to make things compile on my workstation,

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make sure we check to see if the configure script has been executed on a new checkout or after a distclean

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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines

Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
      mixmonitor.patch uploaded by reformed (license 330)

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r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines

when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it

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