[asterisk-dev] Asterisk 1.10 Update

[Mark Murawski]

On 02/17/11 22:35, Dan Austin wrote:
> Russell wrote:
>
>> We would like to release Asterisk 1.10 roughly a year after Asterisk
>> 1.8.  This will be a standard release, not LTS [2].  To have the release
>> out in the October time frame, we need to branch off 1.10 (feature
>> freeze) at the end of June.  At that point we will begin the beta and RC
>> process.  If you're working on new development projects that you would
>> like to get into Asterisk 1.10, please keep this timeline in mind.
>

I hope to have the following items ready for the 1.10 commit deadline 
(whenever it may officially be).

In no particular order:

PreDial - code written and tested
   https://reviewboard.asterisk.org/r/1229
   Say SIP/abc is calling SIP/def
   You have: Dial(SIP/abc)
   SIP/abc-123234 is created.  But how can you tell that from dialplan?
   You can use a pickup macro: M or U options to Dial(), but you have to 
wait till pickup to know.
   PreDial new option 'I' to Dial(), will let you run dialplan on the 
newly created channel before it is connected to the end-device.

   New way:
   Dial(SIP/def,,I(predial,s,1))
   Dialplan will run on SIP/def-123234 and allow you to know right away 
what channel will be used, and you can set specific variables on that 
channel.


Group Variables - code written and tested - needs some minor reworks
   https://reviewboard.asterisk.org/r/464/

   Example:
   SIP/abc-123 sets GROUP()=foo
   SIP/def-456 sets GROUP()=foo

   SIP/abc-123 sets GROUP_VAR(foo,myvar)=testing123
   SIP/def-456 can read ${GROUP_VAR(foo,myvar)}

   Group variables are shared for all members of the group.
   Group variables are destroyed when all members of the group are hung up
   No need for global variables and manual cleanup for keeping data for 
related channels


Dynamic event filtering for AMI - code written and tested
  https://reviewboard.asterisk.org/r/1228/
  https://issues.asterisk.org/view.php?id=19352
  Add new AMI event filters through manager commands

Meetme Action Announcements
  - Announce to users when they have been muted - code written and tested
   https://reviewboard.asterisk.org/r/1010/

  - Additional information for meetmejoin
   https://issues.asterisk.org/view.php?id=16616
   More information like uniqueid of the caller when recieving a MeetmeJoin


  - Announce to users when the conference is locked - code partially written
   - reviewboard soon
   https://issues.asterisk.org/view.php?id=18786


SIP Transfer event improvements - code written and tested
   - reviewboard soon
   In order to know what happened to a channel after a sip attended 
transfer you need to watch for the Masquerade and also the Transfer 
event.  I've added a new event that puts all the information together so 
you can see exactly what happened during a transfer by looking at one event.

Minor improvements to console logging - code written and tested
   - reviewboard soon
   Current console logging for certain events that are going on with 
respect to channels are a bit lacking in context.  I've made several 
changes to log output messages to show relevant context.

Improved dialplan and console Originate - concept
   channel originate and the Originate() dialplan application cannot 
pass dialplan variables.  I plan to add support for this.

Channel Hangup Handlers - code written and tested
   https://reviewboard.asterisk.org/r/1230/
   An alternative and much more dynamic way to run dialplan on channel 
hangup.  An alternative to the 'h' extension.