Noticia de ultima hora: El primer Branck de Asterisk 1.10 ha salido a la luz :)

Bueno apenas nos vamos acoplando a todas las maravillas que ha agregado Asterisk 1.8, vemos que en Venture Voip han publicado una nota de que ya salió el primer branch de Asterisk 1.10, aquí la nota en ingles

At 8:34am New Zealand time this morning a branch named 1.10 was created.



The commit message reads:



Create branch for Asterisk 1.10

 

You can find the branch here:



http://svn.digium.com/svn/asterisk/branches/1.10/



Update:



The UPGRADE.txt document contains:



===========================================================

===

=== Information for upgrading between Asterisk versions

===

=== These files document all the changes that MUST be taken

=== into account when upgrading between the Asterisk

=== versions listed below. These changes may require that

=== you modify your configuration files, dialplan or (in

=== some cases) source code if you have your own Asterisk

=== modules or patches. These files also include advance

=== notice of any functionality that has been marked as

=== 'deprecated' and may be removed in a future release,

=== along with the suggested replacement functionality.

===

=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2

=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4

=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6

=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8

===

===========================================================



From 1.8 to 1.10:



cel_pgsql:

- This module now expects an 'extra' column in the database for data added using the CELGenUserEvent() application.



ConfBridge

- ConfBridge's dialplan arguments have changed and are not backwards compatible.



File Interpreters

- The format interpreter formats/format_sln16.c for the file extension '.sln16' has been removed. The '.sln16' file interpreter now exists in the formats/format_sln.c module along with new support for sln12, sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.



HTTP:

- A bindaddr must be specified in order for the HTTP server to run. Previous versions would default to 0.0.0.0 if no bindaddr was specified.



Gtalk:

- The default value for 'context' and 'parkinglots' in gtalk.conf has been changed to 'default', previously they were empty.



chan_dahdi:

- The mohinterpret=passthrough setting is deprecated in favor of moh_signaling=notify.



pbx_lua:

- Execution no longer continues after applications that do dialplan jumps (such as app.goto). Now when an application such as app.goto() is called, control is returned back to the pbx engine and the current extension function stops executing.

- the autoservice now defaults to being on by default

- autoservice_start() and autoservice_start() no longer return a value.



Queue:

- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members

- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.



Asterisk Database:

- The internal Asterisk database has been switched from Berkeley DB 1.86 to SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3 utility in the UTILS section of menuselect. If an existing astdb is found and no astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will convert an existing astdb to the SQLite3 version automatically at runtime.

Fuente: Venture Voip Blog