asterisk announce - Jul 2010 - Asterisk 1.8.0-beta1 is Now Available!

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.

This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8
testing process.  Please report any issues found to the issue tracker,
http://issues.asterisk.org/.  It is also very useful to hear successful test
reports.  Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

     * Secure RTP
     * IPv6 Support
     * Connected Party Identification Support
     * Calendaring Integration
     * A new call logging system, Channel Event Logging (CEL)
     * Distributed Device State using Jabber/XMPP PubSub
     * Call Completion Supplementary Services support
     * Advice of Charge support
     * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta1

Thank you for your continued support of Asterisk!

At 02:58 PM 7/23/2010, you wrote:
>The Asterisk Development Team has announced the release of Asterisk 
>1.8.0-beta1.
So being the brave type, I downloaded and installed this onto my 
Asterisk Box. Compiled fine and installed fine, but it didn't work.

I kept getting errors on gosub and none of my DAHDI channels were 
visible. So I went back to 1.6.2.11-beta one and all was well again.

Is there something really basic I missed to get 1.8 to work?

Ira

On Fri, Jul 23, 2010 at 8:18 PM, Ira <ira at extrasensory.com>
wrote:
> Is there something really basic I missed to get 1.8 to work?
>
Rather then tell us it did not work, post a debug log showing the issue.

A side from that did you read the UPGRADE.txt and CHANGES file located
in the source directory?

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

On Fri, Jul 23, 2010 at 5:58 PM, Asterisk Development Team
<asteriskteam at digium.com> wrote:
> All interested users of Asterisk are encouraged to participate in the 1.8
> testing process. ?Please report any issues found to the issue tracker,
> http://issues.asterisk.org/. ?It is also very useful to hear successful
test
> reports. ?Please post those to the asterisk-dev mailing list.
>
Remember, when reporting issue to the tracker[1] please take a moment
to first read the bug guidelines[2].  Also be sure to upload a
complete debug log[3] reproducing your issue.

If you believe the issue is a regression (worked with a previous
version of Asterisk), if possible include a debug log from the working
version.

[1] https://issues.asterisk.org/
[2] http://www.asterisk.org/developers/bug-guidelines
[3]
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

At 02:58 PM 7/23/2010, you wrote:
>The Asterisk Development Team has announced the release of Asterisk 
>1.8.0-beta1.
>This release marks the beginning of the testing process for the 
>eventual release
>of Asterisk 1.8.0.
One more problem. Everything seems to work fine but this morning I 
decided to test something. Picked up my SIP phone and tried to call 
myself and it doesn't work.

Phone is an Aastra 480i. I can dial out via SIP or POTS via a TDM400. 
All possible options go straight to voicemail. If I call in from my 
cell or from 2 cells at once it usually works fine. When it doesn't 
work, I get 3 pairs of these, I assume one for each of the SIP phones 
in the house.

WARNING[14583]: chan_sip.c:3339 retrans_pkt: Retransmission timeout 
reached on transmission 
10842037066464ef58d4f88d16535b4c at 192.168.233.235:5060 for seqno 102 
(Critical Request) -- See doc/sip-retransmit.txt.
Packet timed out after 6400ms with no response
WARNING[14583]: chan_sip.c:3368 retrans_pkt: Hanging up call 
10842037066464ef58d4f88d16535b4c at 192.168.233.235:5060 - no reply to 
our critical packet (see doc/sip-retransmit.txt).

It the same dial line in extensions.conf whether it works or not.

I did in fact read doc/sip-retransmit.txt, but it didn't seem to 
contain anything I understood.

I assume this should also be in the bug tracker?

Ira

On Sun, Jul 25, 2010 at 2:53 PM, Ira <ira at extrasensory.com>
wrote:
> I assume this should also be in the bug tracker?
>
A wild stab in the dark, did you Answer() or Progress() before you
called Dial()?  If not, can you add it to your dialplan and retest.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

At 12:53 PM 7/25/2010, you wrote:
>A wild stab in the dark, did you Answer() or Progress() before you
>called Dial()?  If not, can you add it to your dialplan and retest.
Just added progress with no change.

Ira