Great! tnx matt!<br><br><div><span class="gmail_quote">On 11/11/05, <b class="gmail_sendername">Matt Florell</b> <<a href="mailto:astmattf@gmail.com">astmattf@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
It's CVS v1-0. Digium has said that they will do a release of 1.0.10<br>at the same time they release 1.2.<br><br>I highly recommend upgrading to this if you are still on the 1.0 tree.<br>It has a lot of bug fixes, and the new v2 firmware telco cards from
<br>Digium run much better on it than they do on 1.0.9.<br><br>If you want it now, just checkout from CVS like this:<br>cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds<br><br>MATT---<br><br><br>
On 11/10/05, Mark Quitoriano <<a href="mailto:markquitoriano@gmail.com">markquitoriano@gmail.com</a>> wrote:<br>> in the Changelog on<br>> <a href="http://ftp.digium.com/pub/asterisk/ChangeLog">http://ftp.digium.com/pub/asterisk/ChangeLog
</a> there's a<br>> asterisk 1.0.10 which i can't find anywhere, any hints?<br>><br>><br>> --snip from ChangeLog--<br>> Asterisk 1.0.10<br>><br>> -- chan_local<br>> -- In releases 1.0.8 and 1.0.9
, the Local channels that are created would<br>> not be masqueraded into the new channel type. This has now been fixed.<br>> -- chan_sip<br>><br>> -- The 'insecure' options have been changed to support matching peersby IP
<br>> only, not requiring authentication on incoming invites, or both. Before,<br>> to not require authentication on incoming invites also required matching<br>><br>> peers based on IP only.<br>> -- chan_zap
<br>> -- Before, call waiting could occur during the initial ringing on the line.<br>> This has now been fixed.<br>> -- app_disa<br>> -- We will now not set the accountcode if one is not supplied.<br>><br>
> -- app_meetme<br>> -- If the first caller into a conference hangs up while being prompted for<br>> the conference pin number, the conference will no longer be held open.<br>> -- app_userevent<br>> -- Events created with this application were indicated as a "call" event
<br>><br>> instead of a "user" event. This made the "user" event permissions<br>> not work correctly.<br>> -- app_voicemail<br>> -- When using the externpass option for voicemail, the password will be
<br>><br>> immediately updated in memory as well, instead of having to wait for<br>> the next time the configuration is reloaded.<br>> -- app_zapras<br>> -- We now ensure buffer policy is restored after RAS is done with a
<br>> channel.<br>><br>> This could cause audio problems on the channel after zapras is done<br>> with it.<br>> -- res_agi<br>> -- We now unmask the SIGHUP signal before executing an AGI script. This
<br>> fixes problems where some AGI scripts would continue running long after<br>><br>> the call is over.<br>> -- extensions<br>> -- A potential crash has been fixed when calling LEN() to get the length of
<br>> a string that was 80 characters or larger.<br>> -- logger<br>> -- The Asterisk logger will automatically detect when a log file needs to<br>><br>> be rotated. However, this feature could put Asterisk in a nasty loop
<br>> that would result in a crash.<br>> -- general<br>> -- Added man pages for astgenkey, autosupport, and safe_asterisk<br>> --end of snip--<br>><br>> --<br>> Regards,<br>> Mark Quitoriano, CCNA
<br>> <a href="http://www.atamanetworks.com">http://www.atamanetworks.com</a><br>><br>> Fan the flame...<br>> <a href="http://www.spreadfirefox.com/?q=user/register&r=19441">http://www.spreadfirefox.com/?q=user/register&r=19441
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</a><br></blockquote></div><br><br clear="all"><br>-- <br>Regards,<br>Mark Quitoriano, CCNA<br><a href="http://www.atamanetworks.com">http://www.atamanetworks.com</a><br><br>Fan the flame...<br><a href="http://www.spreadfirefox.com/?q=user/register&r=19441">
http://www.spreadfirefox.com/?q=user/register&r=19441</a>