in the Changelog on http://ftp.digium.com/pub/asterisk/ChangeLog there's a
asterisk 1.0.10 which i can't find anywhere, any hints?
--snip from ChangeLog--
Asterisk 1.0.10
-- chan_local
-- In releases 1.0.8 and 1.0.9, the Local channels that are created would
not be masqueraded into the new channel type. This has now been fixed.
-- chan_sip
-- The 'insecure' options have been changed to support matching
peersby IP
only, not requiring authentication on incoming invites, or both. Before,
to not require authentication on incoming invites also required matching
peers based on IP only.
-- chan_zap
-- Before, call waiting could occur during the initial ringing on the line.
This has now been fixed.
-- app_disa
-- We will now not set the accountcode if one is not supplied.
-- app_meetme
-- If the first caller into a conference hangs up while being prompted for
the conference pin number, the conference will no longer be held open.
-- app_userevent
-- Events created with this application were indicated as a "call"
event
instead of a "user" event. This made the "user"
event permissions
not work correctly.
-- app_voicemail
-- When using the externpass option for voicemail, the password will be
immediately updated in memory as well, instead of having to wait for
the next time the configuration is reloaded.
-- app_zapras
-- We now ensure buffer policy is restored after RAS is done with a channel.
This could cause audio problems on the channel after zapras is done
with it.
-- res_agi
-- We now unmask the SIGHUP signal before executing an AGI script. This
fixes problems where some AGI scripts would continue running long after
the call is over.
-- extensions
-- A potential crash has been fixed when calling LEN() to get the length of
a string that was 80 characters or larger.
-- logger
-- The Asterisk logger will automatically detect when a log file needs to
be rotated. However, this feature could put Asterisk in a nasty loop
that would result in a crash.
-- general
-- Added man pages for astgenkey, autosupport, and safe_asterisk
--end of snip--
--
Regards,
Mark Quitoriano, CCNA
http://www.atamanetworks.com
Fan the flame...
http://www.spreadfirefox.com/?q=user/register&r=19441
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It's CVS v1-0. Digium has said that they will do a release of 1.0.10 at the same time they release 1.2. I highly recommend upgrading to this if you are still on the 1.0 tree. It has a lot of bug fixes, and the new v2 firmware telco cards from Digium run much better on it than they do on 1.0.9. If you want it now, just checkout from CVS like this: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds MATT--- On 11/10/05, Mark Quitoriano <markquitoriano@gmail.com> wrote:> in the Changelog on > http://ftp.digium.com/pub/asterisk/ChangeLog there's a > asterisk 1.0.10 which i can't find anywhere, any hints? > > > --snip from ChangeLog-- > Asterisk 1.0.10 > > -- chan_local > -- In releases 1.0.8 and 1.0.9, the Local channels that are created would > not be masqueraded into the new channel type. This has now been fixed. > -- chan_sip > > -- The 'insecure' options have been changed to support matching peersby IP > only, not requiring authentication on incoming invites, or both. Before, > to not require authentication on incoming invites also required matching > > peers based on IP only. > -- chan_zap > -- Before, call waiting could occur during the initial ringing on the line. > This has now been fixed. > -- app_disa > -- We will now not set the accountcode if one is not supplied. > > -- app_meetme > -- If the first caller into a conference hangs up while being prompted for > the conference pin number, the conference will no longer be held open. > -- app_userevent > -- Events created with this application were indicated as a "call" event > > instead of a "user" event. This made the "user" event permissions > not work correctly. > -- app_voicemail > -- When using the externpass option for voicemail, the password will be > > immediately updated in memory as well, instead of having to wait for > the next time the configuration is reloaded. > -- app_zapras > -- We now ensure buffer policy is restored after RAS is done with a > channel. > > This could cause audio problems on the channel after zapras is done > with it. > -- res_agi > -- We now unmask the SIGHUP signal before executing an AGI script. This > fixes problems where some AGI scripts would continue running long after > > the call is over. > -- extensions > -- A potential crash has been fixed when calling LEN() to get the length of > a string that was 80 characters or larger. > -- logger > -- The Asterisk logger will automatically detect when a log file needs to > > be rotated. However, this feature could put Asterisk in a nasty loop > that would result in a crash. > -- general > -- Added man pages for astgenkey, autosupport, and safe_asterisk > --end of snip-- > > -- > Regards, > Mark Quitoriano, CCNA > http://www.atamanetworks.com > > Fan the flame... > http://www.spreadfirefox.com/?q=user/register&r=19441 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
great! tnx matt. On 11/16/05, Matt Riddell <matt.riddell@sineapps.com> wrote:> > Mark Quitoriano wrote: > > you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9 > ? > > Yes, you can read the upgrade.txt file inside the RC2 distribution for > information on the required changes. > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://freevoip.gedameurope.com (Free Asterisk Voip Community) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051116/adc63037/attachment.htm