Asterisk Development Team
2010-Jul-23 21:58 UTC
[asterisk-users] 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 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, 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. 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. 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: 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
Paul Belanger
2010-Jul-24 02:08 UTC
[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
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
Paul Belanger
2010-Jul-24 16:34 UTC
[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
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, > 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] issues.asterisk.org [2] asterisk.org/developers/bug-guidelines [3] 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
Paul Belanger
2010-Jul-25 19:53 UTC
[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
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