On 10/24/07, sammy ominsky <[EMAIL PROTECTED]> wrote:
> Lots of good info to be found at http://www.voip-info.org/

And you can also download the book "Astersiks the feature of
telephony" 2nd edition and read it.
http://www.cyberciti.biz/tips/download-asterisk-howto-tutorial-book.html

It contain mostly up to date information some of it was taken from
voip-info.org.

>
> ---sambo
>
>
> On 24/10/2007, at 08:21, Eran Levy wrote:
>
> > Is there any good guide/tutorial for Asterisk configuration &
> > management?
> > I'm facing many issues/requests and I can't handle it in a short
> > time frame...
> >
> > Eran
> >
> > On 10/24/07, Meir Michanie <[EMAIL PROTECTED]> wrote: I see that
> > posting about asterisk interested many people in the list as
> > I see nobody complained so far.
> > I would like to give my tip from my experience with grandstream ata
> > 488.
> > I bought it from Tikal networks. It came with  a very  old firmware
> > that
> > caused  me a lot of trouble.
> > 1. It would disconnect phonecalls ramdomly.
> > 2. there was no bridge mode between interfaces.
> > 3. very low quality and echo.
> > 4. the caller ID from the PSTN is not forwarded to the sip channel.
> > 5. calling through the FXO did not work as a trunk but I would have to
> > call dial (SIPEXT,timeout,D(pstndest).
> > 6. forced to ring once on the local phone before forwarding call to
> > sip.
> >
> > Today I upgraded to the latest firmware and now all issues beside the
> > caller id were solved. Shame that it didn't come with the latest
> > firmware, I would have a better taste on my mouth but may be lower
> > learning curve.
> >
> > One more post.
> >
> >
> >
> >
> > --
> > Thanks,
> > Eran
>


Ido
-- 
http://ik.homelinux.org/

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