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  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32823">
    <title>Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3,and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32823</link>
    <description>The Asterisk.org development team has released Asterisk versions 1.2.30.3, 
1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 
and 1.6.1-rc2.  These releases are available for immediate download from 
http://downloads.digium.com/.

This update for Asterisk includes a fix for a regression introduced in 
Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the Asterisk 1.6 
branch since release.  All releases with the exception of Asterisk 1.2.30.3 
also contain a vast assortment of bugfixes in these releases.  For a full 
list of changes, see the ChangeLogs:

http://svn.digium.com/view/asterisk/tags/1.2.30.3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.4.23-rc2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.0.2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.1-beta3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.0.1/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.1-rc</description>
    <dc:creator>Asterisk Team</dc:creator>
    <dc:date>2008-12-02T04:58:05</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32822">
    <title>[policy] RCs for 1.6.X.Y releases (Was Re: tilghman:tag autotag_for_asterisk r426 - /tags/autotag_for_asterisk/1.6.0.2/)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32822</link>
    <description>
On Dec 1, 2008, at 5:04 PM, SVN commits to the Digium repositories  
wrote:



I would like to propose that we make release candidates for 1.6.X.Y  
releases.  I consider them equivalent to 1.4.X releases, and we are  
making release candidates for those.  They have already proven helpful  
in the past, and it's not a big deal to make them.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.





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</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-12-02T01:41:50</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32821">
    <title>Re: [Code Review] Patch pbx_dundi.c to periodicallyclean it's database cache</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32821</link>
    <description>


Yes.  The missing / is intentional.  I think you have to leave it off for ast_db_gettree() to work properly.



I will look into this.


- Matthew


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On 2008-11-24 10:00:37, Matthew Nicholson wrote:


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</description>
    <dc:creator>Matthew Nicholson</dc:creator>
    <dc:date>2008-12-01T17:07:31</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32820">
    <title>Re: [policy] Merging between branches</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32820</link>
    <description>
A hardy +1 from me. I have, however, mostly been following the policy of not 
merging 1.4 bugfixes into 1.6.0 since they were not regressions in the 1.6.0 
branch. I say mostly because in some cases the bugs were grievous enough that I 
could not in good conscience exclude the change from 1.6.0 (think crash or 
deadlock bugs).


I agree with this given the slow pace of 1.6.X releases and the fact that moving 
from 1.6.X to 1.6.X+1 introduces potentially unwanted new features and with 
those, potential new bugs.


I mentioned above the slow process that has happened so far with the 1.6.X 
releases. 1.6.0 was released on 2 October. It is now 1 December and there has 
been only one beta of 1.6.1 made so far. Since it must continue through the beta 
cycle and also go through an rc process, I don't see 1.6.1 being released before 
the end of the year. The original plan for the 1.6 series called for releases 
approximately every month. If this were followed, then the number of new 
features in each subsequent rel</description>
    <dc:creator>Mark Michelson</dc:creator>
    <dc:date>2008-12-01T17:06:32</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32819">
    <title>Re: Calendaring and the bounds of Asterisk</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32819</link>
    <description>
On 1 Dec 2008, at 14:44, Russell Bryant wrote:


It seems clear to me - No, in theory,  it should not be in 'core'  
asterisk.

However, it probably isn't practical to implement all the desired  
functionality
outside Asterisk, so given that it looks like useful functionality,  
yes it should be
included.

We need to:
a) ensure we _could_ implement this in a sensible way _outside_
core asterisk by adopting a core API before too long.
b) accept this useful contribution and hope the core doesn't bloat too
much before 'a' happens :-)

(By the way - things can be in the asterisk process without being in the
asterisk core - think 729 codec or chan_skype - that's possible because
they both use well defined APIs)



Tim.


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</description>
    <dc:creator>Tim Panton</dc:creator>
    <dc:date>2008-12-01T15:14:42</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32818">
    <title>Calendaring and the bounds of Asterisk</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32818</link>
    <description>Greetings,

As you all know from the discussions on this list, Terry Wilson has 
worked hard to bring calendar integration to Asterisk.  It is a very 
exciting new feature to be able to make available to our users. 
However, I have heard some concerns about whether this functionality 
makes sense architecturally.  I have thought about this, and would like 
to share my thoughts.

First of all, when I say "the bounds of Asterisk", I am talking about 
the bounds of Asterisk, the application.  I'm talking about the line 
that we must draw that defines what belongs in Asterisk, and what should 
be an external tool that uses an interface provided by Asterisk to 
deliver the functionality.  There is no problem with Asterisk the 
project distributing Asterisk the application, as well as a number of 
other tools that do one thing and do them well.

I would say that in the past, we have put almost everything into 
Asterisk.  However, the discussions about the next generation of 
Asterisk programming interfaces demonst</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-12-01T14:44:15</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32817">
    <title>[policy] Merging between branches</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32817</link>
    <description>Greetings,

Last year, when we first started discussing Asterisk 1.6 branch policy 
[1], we had said that the only time that we would change a 1.6.X branch 
after it had reached a release state is for regressions against a 
previous release.

What that has meant in practice is that as we find and fix long standing 
bugs that apply to all of our releases, 1.6.0 does not receive the fix. 
  Currently, that means a fix would go into 1.4, trunk, and then 1.6.1 
since it is still in beta.

I now believe that this was a mistake.  As it stands today, upgrading 
from 1.4 to 1.6.0 will introduce a number of regressions for users. 
This should never happen (until 1.4 or 1.6.0 is no longer in a supported 
state).  So, I would like to immediately change the policy such that 
every fix that applies to 1.4 should also go into _every_ supported 
1.6.X branch.

We had a discussion about this last week on IRC.  It seemed that 
everyone around at the time was in agreement about the change as far as 
normal bug fixes go.  Howe</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-12-01T13:57:44</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32816">
    <title>Re: [Code Review] manager.c does not compile on OpenBSD anymore. This patch fixes it.</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32816</link>
    <description>
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Ship it!


Looks good.  Put a space after the cast, though.  :-)

- Russell


On 2008-12-01 04:49:32, Michiel van Baak wrote:


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</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-12-01T13:30:58</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32815">
    <title>[Code Review] manager.c does not compile on OpenBSD anymore. This patch fixes it.</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32815</link>
    <description>
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Review request for Asterisk Developers.


Summary
-------

manager.c has a call to ast_channel_alloc with NULL as last argument.
This last argument should be a pointer. And since NULL is not really a pointer compile fails on OpenBSD.

The attached patch fixes it in 1.4
in trunk and the 1.6.X branches I think it's better to fix it by replacing NULL with SENTINEL in this call.


Diffs
-----

  branches/1.4/main/manager.c 159896 

Diff: http://reviewboard.digium.com/r/66/diff


Testing
-------

Compiles and the manager works with this patch attached on OpenBSD 4.4


Thanks,

Michiel


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    <dc:creator>Michiel van Baak</dc:creator>
    <dc:date>2008-12-01T10:49:32</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32813">
    <title>apply_plan_to_number() in chan_zap.c</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32813</link>
    <description>I've been doing some playing with the various ISDN dialplan prefixes that
can be set in zapata.conf, making PRI calls from one asterisk span to
another, and also looking at the relevant code in chan_zap.c / chan_dahdi.c.

I see that when a call SETUP is being processed, apply_plan_to_number()
is invoked for the calling number, the calling ANI and the redirectingnum,
but NOT on the called number.

I was wondering whether there is a reason it was not applied to the called
number, or if this is just an oversight. (I realise that there are very few
situations where the TON supplied by the telco would vary between calls)

It would be nice if the called party number could be canonicalised in the
same way as CLID, ANI and RDINS.

Looking for comparison at the SS7 code in the trunk version of chan_dahdi.c,
it looks like the called number IS processed with ss7_apply_plan_to_number()
so it would probably make sense to add this to the ISDN code too.

Comments?

Cheers
Tony
</description>
    <dc:creator>Tony Mountifield</dc:creator>
    <dc:date>2008-11-28T20:38:47</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32812">
    <title>channels() function in 1.6</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32812</link>
    <description>---cut---
  -= Info about function 'CHANNELS' =-

[Syntax]
CHANNEL([regular expression])
---cut---

Shouldn't the syntax read CHANNEL*S*?

   Philipp Kempgen

</description>
    <dc:creator>Philipp Kempgen</dc:creator>
    <dc:date>2008-11-28T20:18:29</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32811">
    <title>Re: [Code Review] CLI permissions</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32811</link>
    <description>
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(Updated 2008-11-28 13:05:36.768535)


Review request for Asterisk Developers.


Changes
-------

Sorry for all this updates to the review board:
- Make regular expression matching case insensitive.
- Explain that we can use regular expressions inside the configuration sample.


Summary
-------

Implement CLI permissions related to issue 11123.
I would like to receive an architectural review, if this is the right way to implement CLI permissions. I know there are some minor changes that need to be done (and others not so minor), maybe use RWLIST instead of a LIST, also trailing spaces, and a lot more, but If the architecture is ok, I will continue improving the code.

Thanks in advanced!


Diffs (updated)
-----

  /trunk/CHANGES 159772 
  /trunk/configs/cli_permissions.conf.sample PRE-CREATION </description>
    <dc:creator>Eliel Sardañons</dc:creator>
    <dc:date>2008-11-28T19:05:36</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32809">
    <title>Re: [Code Review] Calendaring API for Asterisk</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32809</link>
    <description>
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/trunk/include/asterisk/calendar.h
&lt;http://reviewboard.digium.com/r/58/#comment282&gt;

    Please fill out the rest of the doxygen docs to include return values and arguments



/trunk/main/calendar.c
&lt;http://reviewboard.digium.com/r/58/#comment283&gt;

    Minor nitpick ... copyright is just 2008



/trunk/main/calendar.c
&lt;http://reviewboard.digium.com/r/58/#comment284&gt;

    This list needs to be protected by a lock.



/trunk/main/calendar.c
&lt;http://reviewboard.digium.com/r/58/#comment285&gt;

    It would probably be good to add a comment to make it clear that this line is just for defining a type.



/trunk/main/calendar.c
&lt;http://reviewboard.digium.com/r/58/#comment286&gt;

    This seems like a good candidate for ao2_callback().  Using the new argument, you can pass a variable that the c</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-11-28T01:37:42</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32806">
    <title>Re: FW: CDR save HANGUPCAUSE</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32806</link>
    <description>Sorry, I'm not sure if the mails are actually arriving to the list.



-----Original Message-----
From: asterisk-dev-bounces&lt; at &gt;lists.digium.com
[mailto:asterisk-dev-bounces&lt; at &gt;lists.digium.com] On Behalf Of Steve Howes
Sent: miércoles, 26 de noviembre de 2008 08:52 p.m.
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] FW: CDR save HANGUPCAUSE

Once is usually sufficient.

On 26 Nov 2008, at 22:46, Sebastian wrote:

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database 2911 (20080229) __________

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http://www.eset.com
 
 

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</description>
    <dc:creator>Sebastian</dc:creator>
    <dc:date>2008-11-26T23:25:00</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32805">
    <title>Re: FW: CDR save HANGUPCAUSE</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32805</link>
    <description>Once is usually sufficient.

On 26 Nov 2008, at 22:46, Sebastian wrote:

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</description>
    <dc:creator>Steve Howes</dc:creator>
    <dc:date>2008-11-26T22:51:53</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32804">
    <title>FW: CDR save HANGUPCAUSE</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32804</link>
    <description>_______________________________________________
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    <dc:creator>Sebastian</dc:creator>
    <dc:date>2008-11-26T22:46:07</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32803">
    <title>Re: Difficulty of adding "i" to Page()?</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32803</link>
    <description>
Mark,

Thank you very much, that's excellent.


Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
http://integrics.com/

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</description>
    <dc:creator>Alistair Cunningham</dc:creator>
    <dc:date>2008-11-26T22:28:09</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32802">
    <title>Reviewboard post-commit hook</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32802</link>
    <description>Our "svnbot" functionality has been expanded to reviewboard.  For now, 
I'm using the stock post-commit script distributed by reviewboard.

The current functionality is that if you include the full URL to a 
review in a commit message, the "svnbot" will automatically set the 
review as submitted.  So, when you merge in the final version of a patch 
into the base, include a comment for the review with the full URL, and 
it will be closed for you.

At some point, it would be nice to expand the recognized keywords to be 
more similar to what we use for mantis.  I'll post another message here 
if and when that happens.

</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-11-26T22:13:01</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32801">
    <title>Re: Using ReviewBoard to review work in developer branches (caveats)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32801</link>
    <description>
That seems to sometimes work.  I have a couple of notes, though ...

1) If you're lazy and don't feel like uploading the diff, post-review is 
easier.  The previously mentioned method of merging into a working copy 
and then running "post-review" from there is how you would do that.  I 
would recommend that everyone learn to love post-review, as it is the 
quickest and easiest way to get changes posted, and is generally the 
preferred method to avoid problems such as ...

2) Sometimes raw "svn diff" puts some stuff in the diff that reviewboard 
gets upset about.  post-review has some magic to clean that stuff up 
into a form that reviewboard is happy with, IIRC.

</description>
    <dc:creator>Russell Bryant</dc:creator>
    <dc:date>2008-11-26T21:43:18</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32800">
    <title>Re: Using ReviewBoard to review work in developerbranches (caveats)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32800</link>
    <description>
I've always just done a svn diff https://origsvn.digium.com/svn/asterisk/trunk 
  https://origsvn.digium.com/svn/team/twilson/calendaring &gt; diff.txt,  
then manually removed any configure changes or if anything has snuck  
through because since the last automerge.  Seemed to work fine for  
me...or am I missing something?

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</description>
    <dc:creator>Terry Wilson</dc:creator>
    <dc:date>2008-11-26T21:32:03</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32799">
    <title>Re: Using ReviewBoard to review work in developerbranches (caveats)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/32799</link>
    <description>
26 nov 2008 kl. 22.04 skrev Kevin P. Fleming:


This text should propably be added to the developer section of
Asterisk.org together with other information needed to work
with reviewboard.

Just a suggestion from a cold country far away...

/O

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</description>
    <dc:creator>Johansson Olle E</dc:creator>
    <dc:date>2008-11-26T21:19:00</dc:date>
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