[Asterisk-Users] asterisk 1.0.10?

[Mark Quitoriano]

hey guys,

if i get the asterisk from CVS like "cvs checkout -r v1-0 zaptel libpri
asterisk asterisk-addons asterisk-sounds" do i get a stable one?


On 11/11/05, Mark Quitoriano <markquitoriano at gmail.com> wrote:
>
> Great! tnx matt!
>
> On 11/11/05, Matt Florell <astmattf at gmail.com> wrote:
> >
> > 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 at 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
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>
>
>
> --
> Regards,
> Mark Quitoriano, CCNA
> http://www.atamanetworks.com
>
> Fan the flame...
> http://www.spreadfirefox.com/?q=user/register&r=19441
>



--
Regards,
Mark Quitoriano, CCNA
http://www.atamanetworks.com

Fan the flame...
http://www.spreadfirefox.com/?q=user/register&r=19441
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