[asterisk-dev] Asterisk 1.8 Branch Creation

[Alec Davis]

Would also like to see https://issues.asterisk.org/view.php?id=17318 make
it.

Asterisk when used as a VOIP gateway with sip trunks to the Telco and FXS
analog trunks to a legacy PBX, can only indicate a remote hangup with the
disconnected tone, this adds support for analog FXS line polarity switching
for remote Answer and Hangup call supervison.

Alec Davis

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Russell Bryant
Sent: Thursday, 22 July 2010 11:03 a.m.
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] Asterisk 1.8 Branch Creation

Greetings,

As you may have seen in a comment in another message, it is the intention to
have the 1.8 branch created by the end of the week (likely on Friday).  IPv6
support was the last major pending project, and the final related patches
are up on reviewboard.  We need to get the testing process going on 1.8 as
soon as possible so we can move forward toward the official release.

Ideally, all features that will be in 1.8 should be in trunk by the time the
branch is created.  However, while we're still early in the beta process, I
think it is reasonable to consider making some exceptions for some last
minute additions on a case by case basis.

If you have anything pending, please try to get it in as soon as possible.
If you have something that can't make it by Friday, but you think could be
done in the next couple of weeks, then that's still a possibility.  Let's
talk about it.

Thanks,

--
Russell Bryant
Digium, Inc.  |  Engineering Manager, Open Source Software
445 Jan Davis Drive NW   -    Huntsville, AL 35806  -  USA
jabber: rbryant at digium.com    -=-    skype: russell-bryant
www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org



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