Asterisk 1.10 Update

I just posted an update on the development of Asterisk 1.10 to the asterisk-dev mailing list. Here is the content:

Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting
and discussed some of the projects that people would like to see in
Asterisk 1.10 [1]. We discussed the schedule there a bit, as well. Now
that Asterisk 1.8 has settled down and we are well into the development
cycle for Asterisk 1.10, it is a good time to revisit the plans for the
next release.

At Digium, the biggest thing we have been working on for 1.10 so far is
replacing the media infrastructure in Asterisk. Most of the critical
and invasive plumbing work is done and has been merged into trunk. Next
we’re looking at building up some features on top of that, such as
adding more codecs, enhancing ConfBridge() to support additional
sampling rates (HD conferencing), adding features that exist in
MeetMe() but not ConfBridge(), and enhancing codec negotiation.

Of course, many others have been working on new developments as well. I
would encourage you to respond if you’d like to provide an update on
some new things that you’re working on.

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.

As always, comments and questions are welcome.

Thanks,

[1] https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions


Russell Bryant

4 thoughts on “Asterisk 1.10 Update

  1. One feature I’ve been waiting for since 1.2 is to see the original callerID when a call is transferred.

    For ex: Asterisk receives a call from 310-123-1234, and is picked up by SIP/555. Then it gets blind-transferred to SIP/666. The phone on SIP/666 will show the original callerID as “555”… and not “310-123-1234” which is now lost… I wish we could get this ‘fixed’ 😉

    Can we pretty please see this improvement in the next Asterisk release ?

    Thanks again for an AWESOME project.

  2. Hi there ive pushed my projects onto review board.

    1)some additional options/features in app_queue that i have been using for a while and have put in for users we service
    2)changes to chan_h323 to work better with gatekeepers and latest h323plus
    3)modified app_directed_pickup to run a macro on bridge to “fixup” in the dialplan
    4)T.38 T.30 gateway implementation into res_fax.c/res_fax_spandsp.c

    1-3 are mostly done

    4 im working on have it working but not happy with the T.38 param exchange will be done in week or two … its based on the patch i worked with in 2008 on bug tracker have a big requirement for this at the moment and should be polished by june think it will be a nice fit for *.

    Kind regards Greg

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