[Asterisk-video] [Fwd: [asterisk-dev] Asterisk 1.6.0branch created]

[Sergio Garcia Murillo]

Yes I did, but still have not got any free time to take a look at it (and the rest of bugs).
I hope to get to normality in a couple of weeks, so please be patient.. :)

Best regards
Sergio

----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
To: asterisk-video at lists.digium.com
Sent: Tue, 04 Mar 2008 22:14:08 +0100
Subject: Re: [Asterisk-video] [Fwd: [asterisk-dev] Asterisk 1.6.0branch	created]

btw: have you received my email with dumps of successful/unsuccessful 
h324m calls? Did you had the time to take a look on it?

regards
klaus

Sergio Garcia Murillo schrieb:
> Hi Klaus,
> 
> I'd wait until 1.6.1 to let things stabilize a bit more and to let me finish the rest of 
> the projects I have pending so I can get back to work on this again.. :)
> 
> Best regards
> Sergio
> 
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> To: asterisk-video at lists.digium.com
> Sent: Tue, 04 Mar 2008 19:59:36 +0100
> Subject: [Asterisk-video] [Fwd: [asterisk-dev] Asterisk 1.6.0 branch created]
> 
> I think we should port h324m and other stuff/patches to 1.6
> 
> regards
> klaus
> 
> 
> -------- Original-Nachricht --------
> Betreff: [asterisk-dev] Asterisk 1.6.0 branch created
> Datum: Tue, 04 Mar 2008 11:07:00 -0600
> Von: Russell Bryant <russell at digium.com>
> Antwort an: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> An: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> 
> Greetings,
> 
> I just created a 1.6.0 branch in svn.  What does this mean?
> 
> 1) Asterisk 1.6.0 is now feature frozen and nearing release candidate 
> status.  I
> have two issues against Asterisk 1.6 that I would like to resolve first:
>     - 12130, G.722 transcoding problems
>     - 11972, deadlocks related to SIP TLS
> 
> 2) Asterisk trunk is now completely open for changes.  There are 
> multiple heavy
> sets of changes that have been waiting to get merged until 1.6.0 is 
> done.  They
> can now be merged in to trunk.
> 
> 3) If you are a committer to the Asterisk repositories, you have 1 more 
> place to
> merge bug fixes.
> 
>     a) Merge fix into 1.4 as usual
>     b) Merge up trunk as usual
>     c) Merge the bug fix from trunk to 1.6.0 using the "mergetrunk6" wrapper
>        that is in repotools.
> 
> That's all for now.  Questions or comments are welcome, as always.
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-video