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r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 lines

Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)

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r57870 | file | 2007-03-05 12:52:03 -0500 (Mon, 05 Mar 2007) | 10 lines

Merged revisions 57869 via svnmerge from 
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines

Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)

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r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon, 05 Mar 2007) | 9 lines

Merged revisions 57825 via svnmerge from 
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line

Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2 lines

Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)

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r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2 lines

Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)

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r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines

Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)

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the conversion of IAX2 variables from the current custom method to ast_storage.


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r57649 | tilghman | 2007-03-03 00:45:00 -0600 (Sat, 03 Mar 2007) | 10 lines

Merged revisions 57648 via svnmerge from 
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines

Memory leak of a list, if call recording was abandoned

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r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007) | 1 line

submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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Thanks to Lacy Moore on the asterisk-users list for pointing out that this
was missing!


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r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02 Mar 2007) | 3 lines

Update the check that is used to determine whether zaptel transcoder support
is present.  The interface has changed.

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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines

Merged revisions 57475 via svnmerge from 
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines

If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.

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r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri, 02 Mar 2007) | 9 lines

Merged revisions 57458 via svnmerge from 
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line

further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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application.  (issue #9187, Junky)


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r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1 line

I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL.
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r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) | 4 lines

Return the correct digit that interrupted the stream.  This fixes exiting the
Background application when using the m option.
(issue #9176, mjagdis)

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r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines

Merge changes from svn/asterisk/team/russell/sla_updates

* Originally, I put in the documentation that only Zap interfaces would be
  supported on the trunk side.  However, after a discussion with Qwell, we came
  up with a way to make IP trunks work as well, using some things already in
  Asterisk.  So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
  voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
  in SLA.  The station's channel needs to be passed to the dial API when
  dialing the trunk.
* Change a WARNING message to DEBUG in channel.h.  This message is of no use
  to users.

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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines

Merged revisions 57317 via svnmerge from 
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines

Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)

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r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | 2 lines

minor tweaks to the sla docs

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r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines

Merge more changes from svn/asterisk/team/russell/sla_updates

* Add support for private hold.  By setting "hold=private" for a trunk, only
  the station that put the call on hold will be able to retrieve it from hold.
  Also, by setting "hold=private" for a station, any call that station puts
  on hold can only be retrieved by that station.

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Convert the PBX core to use read/write locks. This yields a nifty performance improvement when it comes to simultaneous calls going through the dialplan. Using murf's test the old mutex based core took an average of 57.3 seconds while the rwlock based core took 31.1 seconds. That's a nifty 26.2 seconds performance improvement. The other good part is that if we do need to switch back then we just have to change the lock/unlock API calls. I converted everywhere that used to touch the mutex locks directly to use them.


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r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) | 2 lines

Minor formatting change

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r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines

Merge changes from svn/asterisk/team/russell/sla_updates

* Add support for the "barge=no" option for trunks.  If this option is set,
  then stations will not be able to join in on a call that is on progress
  on this trunk.

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r57139 | murf | 2007-02-28 12:23:05 -0700 (Wed, 28 Feb 2007) | 9 lines

Merged revisions 57118 via svnmerge from 
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r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line

a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails
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r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines

Merged revisions 57092 via svnmerge from 
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines

Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)

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r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | 8 lines

Merge current set of changes from svn/asterisk/team/russell/sla_updates

* Add support for station ring delays.  Ring delays can be set globally for a
  station or for specific trunks on the station.
* Fix a few bugs in existing code.
* Restructure and Reorganize code to improve readability and maintainability.
* Improve formatting of the "sla show (trunks|stations)" CLI commands.

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r57055 | file | 2007-02-28 12:55:03 -0500 (Wed, 28 Feb 2007) | 2 lines

Picky compiler...

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r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb 2007) | 2 lines

Better handle timeouts when the individual speaks after everything has been played but before the timeout ends.

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r57049 | murf | 2007-02-28 10:15:27 -0700 (Wed, 28 Feb 2007) | 1 line

I was surprised that I had not yet downgraded missing goto targets and macro call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code.
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Properly hangup the original dialed channel, not the new channel that appeared from the forwarding. (issue #9161 reported by PhilSmith)


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Merged revisions 56975 via svnmerge from 
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r56975 | russell | 2007-02-27 11:36:09 -0600 (Tue, 27 Feb 2007) | 4 lines

Fix voicemail email attachments.  I missed the conversion of one of the line
endings and there was an extra one where it should not have been.
(issue #9128)

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