https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58933 | tilghman | 2007-03-15 17:32:33 -0500 (Thu, 15 Mar 2007) | 2 lines

Function works fine, but the documentation is backwards.

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r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines

Merge changes from svn/asterisk/team/russell/LaTeX_docs.

* Convert most of the doc directory into a single LaTeX formatted document
  so that we can generate a PDF, HTML, or other formats from this
  information.
* Add a CLI command to dump the application documentation into LaTeX format
  which will only be include if the configure script is run with 
  --enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included.  However, you
  can simply run "make asterisk.pdf" to generate it yourself.  We may include
  it in release tarballs or have automatically generated ones on the web site,
  but that has yet to be decided.

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r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines

Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)

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r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines

Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration.  While we're at it, support "limitonpeerz" and 
"limitonpeerssssss".  (inspired by issue #9172)

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r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14 Mar 2007) | 2 lines

Add a more basic example setup to the examples section

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THanks Philippe - Greetings from Lisboa, Portugal


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r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines

Merged revisions 58847 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)

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r58897 | russell | 2007-03-14 11:40:22 -0500 (Wed, 14 Mar 2007) | 11 lines

Merged revisions 58896 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines

Add a note to the security file that the Asterisk CLI and log files may contain
sensitive information, and that people should keep this in mind.

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r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines

By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations.  However, add an option to
enable it for those that would like to use it anyway.

The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.

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r58880 | tilghman | 2007-03-13 20:47:08 -0500 (Tue, 13 Mar 2007) | 3 lines

Issue 9162 - pbx_substitute_variables_helper assumes the buffer is initialized
to all zeroes.  This fixes a case where it wasn't.

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r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) | 4 lines

Ensure that the blinky lights show that the trunk stopped ringing when the
trunk hangs up before a station has answered it.
(issue #9234, reported by francesco_r)

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r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 Mar 2007) | 1 line

fix the reference to the SLA documentation
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
  SQLite3 database.  (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
  support for SQLite version 2.  I decided that this was ok since we didn't have
  any realtime support for version 3.  If someone ports this to version 3, then
  version 2 support can be removed or marked deprecated.
  (issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.

Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality.  Those are:

* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)


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r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines

Don't hangup the call on OK or errors on MESSAGE and INFO 
inside of a dialog (like video update requests).

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r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9251 - Clear From URI from user attributes (tgrman)

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r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines

Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)

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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines

Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)

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 for Mark Spencers upcoming 30th birthday.

To enable, run `make menuselect` and select the option MARKO_BDAY under Compiler Flags.


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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines

Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
 - Interpolating a frame when the jitterbuffer is in use
 - decrypting a frame when IAX2 encryption is on
 - frames in an IAX2 trunk

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r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) | 2 lines

Make the compiler happy and initialize a variable.

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r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) | 8 lines

Merge some updates to the SLA documentation.  I plan to keep working on this
to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things.  However, I got tired of doing
it in plain text, so I switched to using LaTeX.  I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.

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r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2 lines

Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai)

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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines

Merged revisions 58579 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines

If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.

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r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | 5 lines

Hang up the channel that put the call on hold in the event processing thread to
avoid a race condition.  Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.

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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines

Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242, andrew)

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r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007) | 2 lines

Fix segfault (Issue 9236)

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r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) | 3 lines

Refactor hold handling a bit so that it does not require keeping the call up
when a call is put on hold.

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r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines

Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)

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r58389 | file | 2007-03-08 11:07:10 -0500 (Thu, 08 Mar 2007) | 10 lines

Merged revisions 58388 via svnmerge from 
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines

Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)

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r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007) | 2 lines

this change was not needed; fclose() handles closing the file descriptor already

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r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007) | 2 lines

fix two cases where HTTP session file descriptors would not be closed

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r58352 | kpfleming | 2007-03-08 08:17:42 -0500 (Thu, 08 Mar 2007) | 2 lines

fix a compiler warning, and overwriting 'res' value

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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines

If we receive ZT_EVENT_REMOVED, destroy the specified channel. 
(issue #7256, tzafrir)

Also, update the configure script to make sure that we don't try to build 
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.

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This can be done using a global variable or a dialplan function.  Using the
SHELL() function will allow you to use an external script to determine what the
weight in the response should be.  This can be very useful in load balancing
applications.
(inspired by discussions with blitzrage and jsmith in #asterisk-bugs)


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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines

(This bug was reported to me by Kinsey Moore)

Merged revisions 58242 via svnmerge from 
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines

Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call.  This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it.  However, this is not always
correct.  So, I have made it change this name after authentication happens
since at that point, we have an exact match.

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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines

Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)

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(issue #9105, junky)


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